arta, steps, limp

ARTA, STEPS, LIMP
A Compendium for the
Programs ARTA family
Base material ARTA Manuals
Dr. Heinrich Weber, German editing and additions
Dr. Ivo Mateljan, English original manuals
© Weber / Mateljan
Version 2.31D, January 2011 (ARTA 1.70)
ARTA - Compendium
Changes Version 2.00D
Chapter
Note to changes / additions
2
3
3.1
5
5.4.2
5.5
6
6.1
7.1
8
8.1.1
8.1.2
11th
11.1
12th
13th
Editorial changes
Modification of the BOM
Supplement the matching calculation
Addition of a reference
NEW: calibration of microphones below 500 Hz
NEW: Review of the measuring amplifier
Addition, modification
NEW: Measurement of the reverberation time
Adaptation to ARTA version 1.1.1
NEW
NEW: Graphical representation
NEW: overlays
NEW: Specifications
NEW: determination of Xmax
Supplement the literature
NEW: Annex with useful tools
Changes in version 2.10D - ARTA 1.4
Chapter
Note to changes / additions
1
3
4
5
6.1
7.1
6.1.2
6.3
7.2
6.3
7.3
7.3
7.3
10.2.1
10.4
NEW: Getting Started with ARTA.
Modification: Calibration with single-channel measurement
Supplement: Assessing the quality of sound cards
Supplement: Fundamental to the calibration of the measuring chain
Supplement: Signal / noise ratio
NEW: a rapid method for the determination of bias (Farina)
Supplement: What are impulse responses from
NEW: Short Cut now available for markers
NEW: Sound Level Meter (ARTA 1.4)
Supplement: Note the value setup for acoustic measurements
NEW: Detecting resonances
NEW: Overlay impulse response (ARTA 1.4)
Supplement: CSD extension to 2048 FFT
Supplement: accurate resistance measurement with cheap multimeter
Supplement: Indication of the quality of measurement cables and terminals
NEW: Voltage or power-related measurements
NEW: Measurement of XLIN
NEW: Grid in spectrogram
NEW: Dual Gate
NEW: Delay Finder in FR2
AN07
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Changes in version 2.20D - ARTA 1.5
Chapter
Note to changes / additions
3
6.2.2
6.4.2
6.6
6.6
7.4
8.4
14
15
Modification: ARTA measurement box
NEW: Automatic calculation of room acoustic parameters
Supplement: Combining reflective membrane channel and in the near field
NEW: Dealing with targets
NEW: Delay for phase-matching
NEW: Create wav files with Arta generator
NEW: Effective use ARTA
NEW: Formulary
NEW: Index
NEW: Octave SPL and Noise Rating
NEW: Third octave SPL and Noise Rating
Changes in version 2.30D - ARTA 1.70
Chapter
Note to changes / additions
0
1.2
3.2
5.2.2
6.0.3
6.0.4
6.2
6.6
6.7
7.1
7.3
8.1.1
8.1.2
9.2
14
Modification: Foreword
NEW: Erfoderliches and useful accessories
Supplement: Single-channel measurements calibrated
Modification: Microphone sensitivity from a near-field measurement
Modification: Averaging
NEW: Excitation signals & "Signal Generation and Recording"
Modification: Measure, but where?
NEW: Frequency and impedance response in a diagram
NEW: Electrical measurements on crossovers
Supplement: Harmonic distortion sine
NEW: Downsampling for the analysis of spatial resonance
NEW: Storing comments in Copy mode
Supplement: Handling of overlays
Supplement: Voltage or power-related measurements with STEPS
Supplement: Formulary
NEW: acoustic model
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Content
0
Foreword
1
1.1
1.2
1.3
1.4
2
Getting Started with ARTA
Conditions for the use and installation ............................................ ......................... 7
Necessary and useful accessories .............................................. .......................................... 7
Pin assignment of NF-NF cables and connectors ......................................... ................................ 12
Measurement setup, basic possibilities .............................................. ..................................... 13
ARTA, the first steps in a hurry ............................................ .................................................. 15
3
3.1
3.2
4
4.1
4.1.1
4.1.2
4.1.3
4.2
5
5.1
5.1.1
5.1.2
5.2
5.2.1
5.2.2
5.2.3
5.3
5.3.1
5.3.2
5.4
6
6.0
6.0.1
6.0.2
6.0.3
6.0.4
6.1
6.2
6.2.0
6.2.1
6.2.2
6.3
6.4
6.4.1
6.4.2
6.5
6.6
6.7
7
7.1
The ARTA MessBox
Calibrated two-channel measurements with the ARTA MessBox .......................................... 21 ......
Single-channel calibrated measurements ............................................... ............................................ 24
Sound card setup and test .............................................. .................................................. 26 ..
Sound Card
Setup WDM driver for Windows 2000 / XP .......................................... ............................ 27
Setup WDM driver for Vista / Windows 7 .......................................... ............................... 29
Setup ASIO
Test of
Calibration
Calibration of the sound card
Calibration
Calibration
Level calibration of the
Enter the sensitivities of the specification ............................................ 46 ..............
Determine the sensitivity of a microphone near field .................................. 47
Tweeter as
Compensate for frequency response errors of the microphone ............................................. 53 ...........
Calibration using a reference-quality microphone> 200 Hz .............................. 55
Calibration below 500 Hz in a pressure chamber ........................................... 57 .............
Testing of the measuring amplifier
Measured with ARTA
Test leads
The signal / noise ratio of the measurement chain ............................................
...................................... 69
Averages
ARTA
Impulse responses - theory and practice ............................................. ......................................... 77
Measure, but where?
Measure under
Determination of the reverberation time of the room characteristics ........................................... 97
............
The automatic evaluation of the reverberation time ............................................. .................. 103
Setup for acoustic measurements on loudspeakers ............................................ 105 ...........
Scaling and joining of near-and far-field measurements .......................................... ............... 114
Closed
Load and
Working with
Electrical measurements on crossovers with ARTA ............................................ 138 ............
Special measurements and examples ............................................... ........................................... 145
Measurement of harmonic distortion sine ............................................ 145 ............
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7.2
7.3
7.4
8
8.1
8.1.1
8.1.2
8.2
8.3
8.4
9
9.1
9.2
9.3
10th
10.1
10.2
10.2.1
10.2.2
10.2.3
10.2.4
10.3
10.4
11th
11.1
12th
Sound level measurements with ARTA ............................................... ...............................................
149
Detection of resonances including downsampling ............................................ ...................... 154
Create wav files to the external excitation signal with ARTA ......................................... 163 .......
Dealing with data, data files, shortcuts, etc. ......................................... ................... 164
Graphical representations in ARTA .............................................. ............................................ 164
Output and formatting of diagrams ............................................. ....................... 164
Working with
Editing data and data files ............................................. ................................. 170
Scale and Scale
Keyboard Shortcuts - ARTA effective use ............................................ ......................... 175
Measure with STEPS
Basic setting of
Amplitude frequency response and distortion measurements with STEPS ....................................... 179
Voltage or power-related measurements with STEPS ........................................... 185 ...........
Measuring with LIMP
Basic setting of
Determination of
13th
Literature
14th
Small Formula and images collection ............................................. ..................................... 211
15th
Index
Notes
Measuring the DC resistance with a cheap multimeter ................................ 199
RLC measurement with
The accuracy of the impedance measurement .............................................. ....................................
202
Recommendations for speakers Specifications ............................................... 203 .................
Determination of deflection XMAX .............................................. .............................................. 205
ARTA Application Notes
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0
Foreword
This compendium is to first-time adopters to use the ARTA program family near
put. It is essentially limited to speaker measurements. The compendium is neither
a translation nor a substitute for the original English manual. It is recommended that the
Original manuals parallel to consult.
An additional source of information, the ARTA website dar. there are for the user
current information and application notes provided.
While it is intended to in the course of time constantly add and update the compendium.
We therefore ask for your understanding if not any time each mask to the current ARTA release
equivalent. Improvements and corrections, and suggestions for program enhancements
are always welcome.
Mitter while the still considerable scope and structure of the compendium is criticized and
a division into separate tutorials desired. Judging by the history of the
Compendium - ultimately by desires of forums and mails as well as the extension of ARTA
externally controlled - it is not surprising that the educational aspect of the passage of time in the
Background came. For lack of time to restructure the moment is not to make.
Perhaps the newly inserted index in the search for topics and answers may
help a little.
The ARTA family will in the foreseeable future on two-or even multilingual menu
converted. For this reason, in this compendium of the English
Terms from the respective menus used to reference to the respective fields and masks
to keep.
The programs of the family currently include ARTA ARTA, STEPS and LIMP. The
Application can be described briefly as follows:
ARTA - Measurement of the impulse response, transfer function and real time analyzer
STEPS - Transfer function, distortion measurements, linearity measurements
LIMP - Impedance measurements on loudspeakers and determination of the TSP
Note: Some of the methods are exclusively for the DIY area
suitable. They are a concession to the limited availability of high quality
Normal.
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1
Getting Started with ARTA
1.1
Conditions for use and installation
Use of the programs of the ARTA family assumes that the following conditions are met
must be:
Operating system: Windows 98 / ME / 2000 / XP / VISTA / Windows 7
Processor: Pentium 400 MHz or higher, Memory 128k
Soundcard: full duplex
The Installing the programs is very simple. Copy the files to a directory and
unzip it then. That's it! All necessary registry entries to be
the first program start automatically saved.
1.2
Necessary and useful accessories
Introduction, a small bill with the necessary and useful accessories, each provided with
first notes and cross-references to more detailed points in the Compendium.
Sound Card
Sound cards can be classified into three groups:
1 Standard sound cards that are on the computer's motherboard (onboard)
2 Additional sound card for PCI or ISA bus (plug-in cards)
3 Sound cards that are connected via a USB or Firewire interface with the computer.
The classes referred to in principle speak to different audiences and different addition of varying quality - by the nature of the connections and, consequently, the type of
necessary cables. For pinout information of common
Connectors and cables, see chapter 1.3.
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Standard sound cards use a stereo cable and 3.5mm jack sockets (Figure 1.2.1).
Semi-pro, high-quality sound cards usually RCA jacks and
unsymmetrical compounds (Figure 1.2.2). Professional sound cards use 6.3 mm stereo
Jacks for balanced connection, 6.3 mm mono jacks for unbalanced
Connections and XLR connectors for balanced microphone ports (Figure 1.2.3).
Standard stereo sound cards have three lines (1, 2, 3) have 5 +1 surround sound systems
three connections (4, 5, 6) on the main board. One of the outputs is designed so that at
his headphones can be operated with a nominal impedance of 32 ohms. Is for testing sound cards
a loopback connection from line-in (blue) to line-out (green) with stereo cable with 3.5mm
Plugs made. The input impedance of the line-in input is at most
PC sound cards 10 to 20 ohms.
1 Line-In / AUX input, stereo (blue)
2 Line Out - Headphone / Front Speaker, Stereo (Green)
3 Mic In - Microphone, mono (pink)
4 Out - Center and subwoofer (Orange)
5 Out - Rear Speakers, Stereo (Black)
6 Out - Side Speakers, Stereo (gray)
Figure 1.2.1: Audio connectors on the motherboard of a PC for a 5 +1 surround sound system
Laptops or notebooks have usually only a stereo headphone output and a
Mono microphone input. Note that this configuration is only with severe restrictions on
Suitable measurement purposes, since - due to the mono input - no measurements in dual channel
Mode and no impedance measurements.
Figure 1.2.2: PCI card with RCA connectors (eg M-Audio Audiophile 24/96).
Examples of plug-in cards are the Basic Terratec 24/96 or the M-Audio Audiophile 24/96 apply. In the
Rule, these cards each have separate RCA connectors for input and output. There
is the left channel and the right channel in white red.
Figure 1.2.3 shows an example of a professional, high quality sound card with firewire
Port. On the front there are two XLR microphone inputs. This input is designed as
Executed combo jack, i.e. in the middle of the XLR connector, a 6.3 mm jack
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can be connected. It serves as an instrument input. The input impedance of the instrument
Input is between 470 ohm and 1 Mohm. Both inputs have a volume control.
Figure 1.2.3: Professional Sound System with Firewire interface
The microphone input can be switched to a 48V phantom power supply of the microphone
are. Furthermore, there is the master volume control to adjust the level of output and
Input monitor and a headphone jack with volume control. On the rear panel are
two balanced inputs, two balanced outputs, SPDIF optical ports and two
Firewire connections.
So far the following sound cards have been used successfully:
RME Fireface 800, RME Fireface 400, RME DIGI96, RME HDSP
Duran Audio D-audio, EMU 1616m EMU 0404 USB EMU Tracker Pre
Gina24 Echo, Echo Audio Fire 4, Echo Layla 24, Echo Indigo
M-Audio Audiophile 2496, FireWire Solo, USB Transit, Delta 44
Basic Terratec 24/96, FW Firewire X24, YAMAHA GO46
Digigram VxPocket 440 - a PCMCIA card
TASCAM US-122 - USB audio
ESI QuataFire 610, July, and U24 USB Waveterminal,
Sound Blaster X-Fi, Infrasonic Quartet
Sound Blaster Live 24, Audigy ZS, Extigy USB (48kHz sampling frequency)
Turtle Beach Pinnacle and Fuji soundcards
With restrictions following sound cards can be used:
Sound Blaster MP3 + USB (Note: Please do not install the SB drivers, they use the
Windows XP default drivers),
Sound cards and on-board audio with AC97 codec (problems with noise in the FFT mode).
For further information about successfully used sound cards can be found on the homepage
of ARTA: http://www.fesb.hr/ ~ Mateljan / arta / index.htm.
Furthermore, the entry is an essay by Marcel Müller: Technical properties of sound cards
the PC [33] and Section 2.1 is recommended.
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Amplifier
In principle, each power amplifier with a linear frequency response and a power> 5 to 10
Watts appropriate. The output resistance RA should be <0.05 ohms. Warning, do not use
Amplifier virtual ground (bridge amplifier) that could get your sound card bad.
If you are not sure what type is your amplifier so you better inquire before
first use the manufacturer. An inexpensive recommendation that meets the above conditions and in addition is a mobile application through small counter dimensions - is the t.amp
PM40C of Thomann (see also Section 5.4).
Microphone
The supply of affordable measurement microphones is manageable. It is essential that the microphone
a linear frequency response and omnidirectional polar pattern (Figure 1.2.4) has. Very cheap
Microphones (Behringer ECM8000 for example) are
a compensation file (see Section 5.3) for
normal development work in the speaker
construction quite useful.
When the microphone at higher levels or even
be used for distortion measurements
is, then, a deeper grip in your wallet
required. Recommendations in the medium price range
(150 - € 300) are the Beyerdynamic MM1
and the Audix TM-1 (see also
Chapter 5.2.1 and 9.2).
Image 1.2.4: Radiation Audix TM1
Furthermore, of course, there is the possibility of Selbstbaues with electret. A purpose geignete
Microphone capsule is the Panasonic WM 61A. Guidelines for construction are in ARTA - Hardware &
Manual tools to find.
Mic (MVV)
Depending on the microphone and / or sound card, different additives are required. When
You have chosen a sound card with integrated MVV and 48V phantom power
then you are complete.
If you only call a "naked" sound card you own, so you will need a separate
MVV. In the event that you e.g. have one of the above "buying microphones", should the
MVV can come up with a phantom power. Here is the recommendation: MPA 102 of
Monacor, currently the only affordable MVV with staged - that is reproducible Level control (see also Figure 3.5).
In the case of DIY microphone, you can either directly to the microphone input of the sound card
use (see also section 1.3) or an MVV kits from the internet. Recommended here
Posts by Ralf Grafe (http://www.mini-cooper-clubman.de/html/hifi_projects.html). There
There are several proven kits, z.T. boards are even available.
ARTA MessBox
The ARTA MessBox is not absolutely necessary, but simplifies the measuring life (see further
Chapter 3 or Application Note No. 1). Also for the ARTA MessBox is a platinum solution at
http://www.mini-cooper-clubman.de/html/hifi_projects.html available.
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Cable
To connect all components mentioned several cables are required. At least when a
missing, you know to appreciate its value. Make sure all connections on quality! Loose connection
contacts, poor shielding, etc., can a life embitter the measurement (see also Section 6.0.1).
In general, the following connections are required:
•Microphone cable (depending on the microphone and mic
XLR, TRS, RCA, see also Figure 1.3.1)
•Cable sound card - MessBox
•Cable amplifier - MessBox
•Cable MessBox - Speakers (1.5 to 2.5 mm ²)
Please make sure that all connections are only as long as necessary!
Other useful tools
Loopback cable (to calibrate the sound card, see Chapter 4)
Voltage divider (for level adjustment, see chapter 5)
Y-cable (for the realization of semi-two-channel measurements, see Chapter 2)
Luster terminals, alligator clips (to make temporary connections)
Multimeter (DMM)
A good multimeter is essential for the calibration of the measuring chain and of course also
a useful tool for measuring everyday. If you do not have multimeter in your toolhave luggage, you should decide multimeter ideally a so-called true RMS. The
Offer is great, also under 100 €, there are already useful devices.
If you already have a DMM, or flirting with a cheaper device, which does not
the o.g. Is assigned to category, then you should use the following test before calibration
perform:
a) Connect the multimeter to the
left line output of the sound card and
set the measuring range to 2 volts AC.
b) Start e.g. of the signal generator
STEPSim menu "Measurement
Setup ".
c) Measure at different Testfrefrequencies between 20 Hz and 1000 Hz
with the multimeter the output voltage
the sound card and record the relevant
Value.
Image 1.2.5: Multimeter Comparison
Then enter the values measured either absolute or relative, depending on the frefrequency to. Picture 1.2.5 shows the result for a high average and for a DMM True
RMS multimeter. You can see that when fed with a sinusoidal signal to the frequency dependence
1000 Hz is less than 2-3%. Thus, the DMM would be for the calibration of ARTA with precycle (500 Hz) (see also Section 5.1.1).
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1.3
Pin assignment of NF-NF cables and connectors
Unbalanced
JACK STEREO
Housing: mass (GROUND / SHIELD)
Tip: Plus (LIFE)
Ring: minus (LIFE)
Symmetrical
XLR
Pin 1: GND (GROUND / SHIELD)
Pin 2: positive (LIFE)
Pin 3: negative (LIFE)
Image: 1.3.1: Pin assignment of connection cables
If you want to get an overview of the range of pre-configured cables, then
is to recommend the "CableGuy" on the homepage of Thomann (www.thomann.de)
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1.4
Measurement setup, basic possibilities
In the original manual and in this compendium is on the following test setups reference
taken:
1
2
3
4
5
6
Single channel measurement setup
Semi-two-channel measurement setup
Two-channel measurement setup
Measurement setup for impedance measurement
Measuring loop (loop back) for sound card testing
Probe (sample)
In the following figures all the listed measurement arrangements are shown schematically.
ARTA nutz the left line output of the sound card to output the test signal and the left
Line input for the detection of the DUT signal (Device Under Test). The right line input is
used as a reference channel.
Acoustic measurements
Figure 2.1: Single-channel measurement
setup
Is in the single-channel
only the signal from the DUT (in this case
Microphone signal via microphone)
detected. The error of the sound card and the
Power amplifiers are in the measurement result
included, they are not compensated.
Figure 2.2: Semi-two-channel measurement
setup
In the semi-dual point is the
right line output as, semi-reference '
be used (error of the sound card
compensated).
Figure 2.3: Two-channel measurement
setup
In the two-channel measurement setup is the
Reference signal at the output of
Tapped power amplifier (error of
Be sound card and power amplifier
compensated).
See also ARTA measurement box in Chapter 3
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Impedance measurements
Image 2.4a: Impedance measurement
with
Power amplifier.
See also ARTA measurement box in Chapter 3
Image 2.4b: the headphone impedance
measurement
Output of the sound card
Note: Headphone outputs of sound cards
not low-usually for the operation
Loads designed!
Test and Calibration
Figure 2.5: Measured loop tests for sound cards
In the measuring loop (loopback)
each line outputs to the line
Inputs connected. The measurement loop
is used for testing the sound card.
See also Chapter 4
Protection of the sound card
Figure 2.6: Sensor (voltage divider)
e.g. -20 DB
R1 = 8k2, R2 = 910
To the input of the soundcard
to protect large voltages is
recommended that a voltage divider to
to use. The probe is shown on the left
an attenuation of 20dB, provided the
Sound card input resistance
10 kOhm has. This protection is in the
ARTA MessBox already realized.
For beginners in the field of metrology, ARTA MessBox is recommended in Chapter 3 to
use.
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2
ARTA, the first steps in a hurry
It is to be understood thoroughly that after installing a program immediately to the crunch to the fairs in this case - to go, but give yourself a chance and ARTA and edit
First this section.
In brief, the issues are addressed, the commissioning of a measurement system with ARTA
or for a single-channel frequency response measurement (see Figure 2.1) or an impedance measurement
(see Figure
Image 2.4a or 2.4b) are observed. Further explanations can be found in the respective
indicated chapters.
Adjust sound mixer
The most common mistake that is made in the quick start is the overdrive the sound card.
Go therefore to the first menu and open the Control Panel Sounds and Audio Devices.
Now take on the sound mixer shown below using the example of an onboard sound card
Setting of recording and output channels before.
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a)
b)
c)
d)
Enable line-in recording mixers.
Set the volume in the recording mixer to almost minimum.
Disable line-in in the output mixer.
Set the volume in the mixer output to almost maximum.
Loopback measurement
Now we are almost ready for the first measurement with ARTA. But Look first at Figure 2.5
and connect inputs and outputs of your sound card in accordance with an appropriate
Loopback cable. Depending on your sound card, you need a different cable (see
Section 1.3).
Now it comes with a small excursion into the Chapter 4.2 further. There, the level of adaptation is
Input and output channels of the sound card described. You also get there via the described
Procedures first indications about the quality of your sound card.
So far so good, but the loopback measurement is only for setting the mixer and the test
the sound card. But you probably want to measure the frequency response of your speakers. What this
missing is a measurement microphone.
If your sound card provides the supply voltage for the microphone, you can use
a simple DIY electret microphone work. So please first check the manual
the sound card, whether to find a reference to the supply voltage (see example below).
M-Audio Transit
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The development of a measurement microphone is very easy. In ARTA Hardware & Tools Manual
for details on the replica.
With the minimum equipment for acoustic measurements (computer with onboard sound card
Power amplifier, measuring microphone) and the above-described basic settings, you can now
the first measurements carried out.
Easy test setup for impedance measurement with LIMP
For impedance measurements onboard sound cards are usually not suitable (see also Chapter
4.1). If you have a sound card with stereo Line IN and a headphone output, use
the measurement setup shown in Figure 2.4b. To measure you only need a 100 ohm
Reference resistor and a little shielded cable.
If your cards have no headphone output, so you can use the following measurement setup.
Depending on the input jack of the sound card (jack or RCA) take on a finished top
Cable and cut off the end you do not need. You also need a banana plug, a
Jack, a 27 ohm (5 W) and two reference resistor 8.2 ohm and 1.0 ohm each ¼
Watt resistors. The instructions for the practical construction can be found in the following image
are.
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Now, before measuring just yet - regardless of the measurement setup - three settings
make:
Define the menu "Measurement Setup" in the
"Measurement Config" the reference channel (right) and the
Reference resistor (27 ohms for example).
Note: The chosen reference resistance should be in
Range between 10 ohms and 47 ohms are and exactly
be known.
Imagine the measurement in the "generator set" the output level so that the
Input channels are not overloaded.
Now you calibrate the system in the "Calibrate Input Channels" menu. To do this, connect the left
and the right channel input of the sound card to the output of the signal generator (Line Out),
perform the calibration with "Calibrate" and leave the menu with "OK".
More information about the measurement of impedance transitions and TSP can be found in Chapter
10
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3
The ARTA Measurement Box
To simplify measurements with ARTA, STEPS and LIMP is recommended the construction of the
ARTA MessBox. It is both for impedance measurements as well as for two-channel frequency response
measurements designed and takes the user from the tedious move the test leads.
For measurements with ARTA measurement box, the mass of input and output of the sound card is not verconnected and thus there are no problems with calibrated measurements with ground loops. For onconstruction see Figure 3.1 to Figure 3.3 and [I] or in ARTA Hardware & Tools Manual [22].
Figure 3.1: The finished ARTA measurement box (left conventional, platinum right
solution)
Figure 3.2: The inner workings of the ARTA measurement box (left conventional, platinum right
solution)
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Note 1
The mass of the power amplifier and the sound card can optionally through a 1k
Separated resistor (R6).
Note 2
Warning, do not use a bridge amplifier virtual ground!
Security
The inputs of the sound card are protected by Zener diodes. The power
amplifier is protected as specified by the manufacturer. Make sure that the
Manufacturer specified nominal impedance does not go below.
Figure 3.3: Schematic of the ARTA
measurement box
Note: In single-channel acoustic measurements, the measuring box is not really necessary.
But if it is used, at least the microphone input should be calibrated.
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3.1
Calibrated two-channel measurements with the ARTA MessBox
For a calibrated measurement of the frequency response with ARTA and STEPS in dual channel mode
should
the gain values are entered for both input channels (Ext. preamp gain). It is
Program defines that the right input channel of the sound card as a reference channel and the left
Channel is used as a measurement channel.
The ARTA MessBox is designed so that it should be suitable for most applications.
If an adjustment of the ARTA MessBox to your own needs is desired is a little
Computational work required.
Below is both the adaptation and the calculation of the "Ext Pream Gain "values for
the "Audio Setup Devices " exemplified with the standard equipment of the MessBox.
Adapting MessBox to the power amplifier (IN line, right)
The resistors R1, R2 form together with the input impedance ZIN of the soundcard
Voltage divider k, by
k= (R2 | | ZIN) / (R1 + R2 | | ZIN)
with (R2 | | ZIN) = R2 * ZIN / (R2 + ZIN)
will be described. That is, the maximum voltage that the power amplifier to the right
Line-in channel of the sound card can be issued is equal to
UMAX = S [V rms] / k
S = input sensitivity of the sound card
Then the maximum power that can be used in the measurement, the same
P MAX = (S [V rms] / k)2 / ZSpeaker
The values chosen in the MessBox for R1 = 8k2, R2 = 910 and a standard values for
ZIN = 10k and an input sensitivity of the sound card = 1V, we calculate the gain
the right input channel (Ext. right preamp gain, see Figure 3.4) following input value
Right Channel = (R2 | | Zin) / (R1 + R2 | | Zin) = (910 | | 10k) / (8k2 + (910 | | 10k)) = 0.0923
at PMAX = 29W for 4 ohm or PMAX = 14.5W for nominal 8 ohm speaker impedance.
If your amplifier this line can not leave or when measured with higher performance
to be, or can the voltage divider must be adjusted accordingly. Is your amplifier
e.g. specified with an output power of 56 watts at 8 ohms and you want the full power
use, the following modifications of the ARTA MessBox are required:
S = k [V rms] / √ PMAX * ZSpeaker = 1V / √ 56W * 8 ohms = 0.0472
with R2 = 910 and ZIN = 10K R1 is to be
R1 = (R2 | | ZIN) / k - (R2 | | ZIN) = 834.1 / 0.0472 to 834.1 = 16837 ohms.
Note: The sensitivity of the sound card is specified in the calibration menu in mVPEAK. The
Adjustment calculation for the measurement box requires VRMS = Vpeak * 0.707.
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Adaptation of MessBox to the mic (Line IN, left)
For the calculation of the gain of the left-channel input (Ext. left preamp gain, see image
3.4) You will need the details of your mic preamp.
In the example shown here, the values of the microphone preamplifier 102 MPA by Monacor be
used (see Figure 3.5):
VMicPreAmp =10 (20dB), output impedance of the mic preamp ZOUT = 100,
R5 = 719, ZIN = 10000
Left channel VMicPreAmp ZIN = * / (ZOUT + R5 + ZIN) = 10 * 10000/10819 = 9,243
The value of R5 is calculated as follows:
R5 = R1 | | R2 - ZOUT = 819-100 = 719
This relationship is derived from the fact that both input channels with the same source impedance
to be operated.
Figure 3.4: Audio menu Device Setup ' for ARTA and STEPS
Note: . The ARTA calibration menu to specify the expected gain (gain) in
absolute values and not in dB. It is calculated to Gain = 10 ^ (dB level / 20).
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VMicPreAmp = 10 ^ (x dB / 20)
20 dB = 10
40 = 100
60 dB = 1000
Figure 3.5: Mic MPA 102 (Monacor)
Note: Here, the specification for the low pass filter does not properly
be, it should read 10.5 kHz.
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3.2
Single-channel measurements calibrated
If you want to perform calibrated measurements in single channel mode, you must also
Enter the gain of the power amplifier (power amplifier gain).
Figure 3.6: Audio menu Device Setup ' for ARTA and STEPS
To measure the gain, either follow the description in Section 5.6 or go
as follows:
1 Measure the level of development in the "single channel mode". Determine and record the
Level with the cursor at 1 kHz.
2 Measure the level of development in the "double channel mode". Determine and record the
Level with the cursor at 1 kHz.
3 Determine the difference between the two measurements and berechenen it "Power Amplifier
Gain "as follows
Power amplifier gain = 10 (difference in level @ 1kHz) /
20)
Note that this approach requires a circuit as shown in Figure 2.2 or Figure 2.3 or a
ARTA MessBox.
Example: From Figure 3.7 the following level values at the cursor can be read at 1 kHz: Single
Channel = 106.21 dB Dual Channel = 96.25 dB. This yields the following level difference calculated =
9.96 dB and Power Amplifier Gain = 10 ^ (9.96 / 20) = 3,148. After entering the value in the field
"Power Amplifier Gain" (see Figure 3.6), should single and dual channel measurements up to the
not corrected frequency response error of sound card and amplifier match.
Please note that this procedure after each change in the gain (volume) of the
Must be repeated power amplifier!
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Figure 3.7: PAG determination of single and dual-channel measurement
Alternatively, you can also use the following, slightly more accurate procedure in dual channel mode FR2:
1 Connect the left channel input with the
selected output channel of the sound card.
2 Connect the right input channel on
a voltage divider with the output of the G
Power amplifier. To measure the
Voltage divider see e.g. Figure 2.6.
3 Enter the absolute value of the voltage divider G
as "Ext right preamp gain "(see Figure 3.6).
4 Set the signal generator to ARTA
"Periodic Noise". To protect the sound card
reduce the output level to about-10dB.
5 Start measuring the FR2 Fashion and note
the value of the amplitude at 1kHz. This measured value
corresponds to the gain of the power amplifier
in dB. The entry in the "Audio
Calculated Devices setup "required absolute
as follows:
Power amplifier gain = 10 (FR level @ 1kHz) / 20)
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4
Sound card setup and test
4.1
Sound Card Setup
Before you start measuring, you have your your sound card and your "hardware"
set up. To do this go to the menu Setup Audio Setup Devices or click the Toolbar
Iconan. Then the dialog box as shown in picture 4.1.1 opens.
Image 4.1.1: Audio Devices setup menu
The "Audio Device Setup" has
in section Soundcard the following controls:
Sound Card driver - Selection of Sound Card Driver (WDM - Windows Multimedia drivers or any of the
installed ASIO drivers).
Input channels - Choice of (stereo) input channels of the sound card. ASIO drivers often have a
greater number of channels.
Output Device - Choice of (stereo) output channels of the sound card. In general, one uses the input and
Output channels of the same sound card (mandatory for ASIO drivers).
Control panel - If a WDM driver is selected, the sound mixer for Windows 2000 / XP or
Open the Sound panel for Vista / Windows 7 Control. If an ASIO driver is selected, opens
the ASIO control panel.
Wave format - For Windows 2000 / XP to the Windows Wave Format: 16 bit, 24 bit, 32 bit or float
be selected. Float means: IEEE floating point single precision 32-bit format. If you use
a high-quality sound card is the wav format 24-bit or 32-bit recommended (Note: Many
Sound cards have been declared as 24 bits, but the true resolution is often less than 16-bit).
For Vista / Windows 7, the float format is recommended. This setting has no effect in the ASIO
Fashion, where the resolution is set in the ASIO control panel.
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in section I / O Interface Amplifier:
LineIn sensitivity - Input sensitivity of the line input in mV peak
Line Out sensitivity - Output sensitivity of the left line output in mV peak
Ext Preamp gain - If an amplifier or a voltage divider in the signal path to the lineInputs is, then you must enter here the gain or attenuation factor.
Otherwise, set the entry to 1
L / R channel diff - Speed difference between the left and right input channel in dB.
Power amplifier gain - If you have connected an amplifier on the line output and one channel
want to measure calibrated, then you must enter here the gain of the amplifier (see section
3.2)
in section Microphone:
Sensitivity - Microphone sensitivity in mV / Pa.
Microphone used - Check box active, it means that the microphone is in use and the graphics in dB re 1 Pa
or in dB re 20 μ Pa is scaled. Using the "ComboBox" is the input channel for the microphone
selected (it is recommended to select the left channel of the soundcard as the microphone input).
The setup data can 'Save Setup " and "Load Setup " be saved and loaded. The
Setup files have the extension '. Cal'
Important note: Please turn on the microphone and line channels at the output of the mixer
Mute sound card, otherwise it could cause feedback in measurements. If you have a
use professional sound card, turn the Direct-or zero-latency monitoring
Line Inputs from.
4.1.1 Setup WDM driver for Windows 2000 / XP
After selecting the sound card sound mixer must be defined in which the inputs and outputs
should be used.
Figure 4.1.2: Selection of input and output channels of the sound card
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For standard sound card, the method is as follows:
1 On the, audio device setup ' the button 'Control Panel ' : This will open the
Windows Menu 'Master Volume ' or, volume control '(see Figure 4.1.2)
2 Select the 'Options' menu 'Properties' the channel of the sound card of the
Output (playback or output) to be used (see Figure 4.1.3)
3 Turn, Line In '(with onboard cards are not always available) and microphone
Input channel 'in the' volume control 'silent (mute or mute)
4 Put the 'Volume Control' and 'Wave' to almost maximum.
5 Select the 'Options' menu 'Properties' the channel of the soundcard as the
Input is used.
6 Select the line or microphone input. When an external microphone
is used, the line-in input should be selected.
7 Set the volume of the line-in input to almost minimum.
Image 4.1.3: Typical structure of a mixer output an onboard sound card on Windows XP
Image 4.1.4: Typical structure of an input mixer, an onboard sound card on Windows XP
Note: Most professional sound cards have their own programs to adapt
Input and output channels or monitoring and control volume via hardware.
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4.1.2 Setup WDM driver for Vista / Windows 7
Microsoft has changed under Vista / Windows 7, the control of "Sound Devices". Now is the
System (sometimes in combination with professional software of the controller
Sound cards) responsible for the basic setting of sampling rate and resolution. The
Operating system to change the default resolution for a high-quality mixing, and possibly also for the
Sample rate conversion to the Float format. It is therefore strongly recommended that the ARTA
Float format to choose and set the sampling rate to the default format. Access is via
the "Windows Sound Control Panel" given. The panel can be reached by clicking the
"Control Panel" in Arta menu, select Audio Device Setup ".
Picture 4.1.5 shows the Vista / Windows 7 Control Panel, which contains four pages. In the first step
the playback side must be set, then the procedure for recording repeatedly
Side.
Image 4.1.5: Vista Sound Control Panel
The setting of the sound card on Vista / Windows 7 runs as follows:
1 Click on channel info to select the playback channel. It is recommended that the
Default audio do not use it as a measuring channel.
2 Click on the "Properties" button opens the "Sound Properties" dialog.
3 Click on the tab "Levels" to open the edition mixer (see Figure 4.1.6).
Turn line-in and mic-mute channels, if any.
4 Click the "Advanced" tab sets the channel resolution and the
Sample rate (see Figure 4.1.7)
5 Repeat steps 1 through 4 for the receiving channel. Select the same sampling
as the playback channel.
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Image 4.1.6: Playback channel properties - output level
Image 4.1.7: Setting the resolution and sampling rate in Vista
Note: There are several drivers that are not stable running with Windows 7. In this case, use
Please - if any - the ASIO drivers for your sound card.
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4.1.3 Setup ASIO driver
Audio Stream Input / Output (ASIO) is a product developed by Steinberg, cross-platform,
multichannel-capable audio transfer protocol. ASIO drivers work separately from the operating system. They
have their own control panel to adjust the resolution and buffer size (Figure 4.1.9). The
Buffer is used for the transmission of the sampled data of the driver to the application program
used. The ASIO control panel is opened by pressing the "Control Panel" button in the
ARTA enable "Audio Device Setup" menu (Figure 4.1.8).
Image 4.1.8: Audio Devices setup ASIO
Image 4.1.9: ASIO control panel to adjust the resolution and buffer size
For music applications, the buffer size is chosen as small as a rule, that just a
stable operation is guaranteed. This results in the lowest latency (system-related delay).
In ARTA latency is not a problem because that is addressed by software. Nevertheless, it is not
recommended to use buffers larger than 2048 samples or less than 256 samples. Some ASIO
Control panel indicate the size of the buffer in samples, while others the size of the buffer in
express ms. In this case, the size of the buffer in the sample with the following expression
be calculated:
buffer_size [samples] = buffer_size [ms] * sample rate [kHz] / number_of_channels.
Some ASIO drivers allow the setup of the buffer size (in samples) that a power of the number 2
are (256, 512, 1024, ....). In this case, the buffer size is automatically set by ARTA.
ARTA always operates with two input channels and two output channels. They are as left and
right stereo channel defined. If ASIO support multi-channel devices, the user must be in the
"Audio Device Setup" menu, select to use the stereo channels (1/2, 3/4, ....).
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4.2
Test the sound card
The easiest way to learn about the quality of their sound card is to use the
, Spectrum Analyzer Mode ' in ARTA. You enter this mode by clicking the SPA
Icon in the toolbar shown above. For the test itself, the following steps
required:
Connect the line inputs of the sound card each to the signal outputs (see next
Picture).
Loopback cable e.g. Pollin stereo audio connection cable, 3.5 mm
Plug to 3.5mm plug. Length of 0.3 m. Order 560 824
derÖffnen in the menu Signal generator setup or use the
symbol
Toolbar of ARTA. First, we are interested in only the red-bordered part of the mask. Provide
The values shown in a:
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Now go to the toolbar and select there the values shown below:
You can also use the menu parameters 'Spectrum Analysis Setup' Set. They enter
This menu Measurement setup.
Select the input channel (input channel)
Left.
Before proceeding, make sure to contact again from the right attitude for your sound
Mixer:
1
2
3
4
Enable Line-In recording the mixer.
Set the volume on the recording mixer to almost minimum.
Disable line-in in the output mixer.
Set the volume level of the mixer to almost maximum.
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In "Spectrum Menu Scaling "
are now shown on the left
Make adjustments.
Start the measurement SPAMode by pressing the Record
Symbols or by pressing
from Run in the menu "Recorder". It
result should be a set
as shown in the following image.
If the signal level is too low
be, as you slowly increase the volume of the Line-In recording mixer until the peak at 1 kHz
has a level of approx-3dBFS (see details of the RMS value of the diagram below).
At the bottom of the chart, the value of frequency and amplitude is shown, on which the
Cursor is positioned. Furthermore, the RMS and THD and THD + N. The cursor itself is a thin line
displayed and can be done by left mouse button or the arrow keys to move left or right
are.
Note: During the measurement, you can in the panel, the parameter type of averaging,
Resetting the averaging counter sample frequency, type of excitation signal and the FFT length
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changed.
To show the area in which move popular sound cards are in the following three
shown different results.
M-Audio Transit
THD + N = 0.0069%
Realtek AC97 Audio
THD + N = 0.1845%
Intel onboard card
THD + N = 0.0858%
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How are the results to be interpreted? What can we in the usability of their own
Sound Card derive from it? As a guideline for the assessment of your sound card like the next
two above are:
If THD + N is less than 0.1%, then you have a suitable sound card.
If THD + N is less than 0.01%, then you have a good sound card.
To check how the frequency response of your sound card looks like, go to the
Measurement mode IMP. Use the Single Point Mode (checkbox , Dual Channel
measurement mode ' empty).
Check by pressing the 'Generate' is whether the line-in of the sound card controls. The
Modulation of the map is indicated by the peak level meter. Unless you have a
have red or yellow, you reduce output Volume ' until all green
's. Now, press, Record 'and wait until the measurement is completed (the peak level meter shows
no more) rash. Press "OK" and you should see something like the following picture.
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Now press
and it appears the frequency response of your sound card
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If your sound card is of good quality, you should see a straight line. Make
However, the resolution of your measurement chart. You can change the settings of the
Change chart by by, Fit 'automatically the upper limit of their Y-axis
Find or search manually by the two arrows on the left next to your setting. The
Measuring range can using the arrow buttons to the left, range 'in the same way
can be set.
Another option you have 'set'. If you press 'Set', the following menu appears:
In the menu, setup graph 'you can set all the essential parameters for the graphics. Come
Returning to the frequency response of the sound card. In heavily splayed representation of the Y-axis (2dB)
we can see more details from the frequency response. We have a variation of about + / - 0.1 dB for
the measured M-Audio Transit USB sound card.
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In the following picture you can see the frequency response of the above mentioned sound
cards.
M-Audio Transit
Line-In
+ / - 0.1dB (20Hz to 20kHz)
Realtek AC97 Audio
Microphone input
+ / - 2.5dB (20Hz to 20kHz)
Intel onboard card
Microphone input
+ / - 6.5dB (20Hz to 20kHz)
For measurement purposes should have a sound card, a lower cut-off frequency (-3 dB) of at least 10 Hz or
better have 5 Hz. The ripple (ripple) frequency response should be in the range of 20 Hz - 20000
Hz does not exceed 0.5 dB.
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For assessing the quality of sound cards whose intrinsic noise is also important, which is considered
especially for measurements with noise signals.
To illustrate the impact of a high noise level in the following example:
Of the sound card shown above, the Realtek a noise level of about -80 dBFS
at 20 Hz, the M-Audio Transit about -120 dBFS.
We adopted to encourage our speakers with MLS or white noise and select a
FFT sequence of N = 32768 values. This sequence has N / 2 = 16384 spectral components with a
Power of P = 10 * log (1/16384) = - 42dB under RMS level.
Furthermore, to take into account the crest factor of the excitation signals of about 10-11 dB at
white noise and 6-9 dB at MLS.
Note: The crest factor is the ratio between peak and
Rms value of an alternating quantity (Cf = U.S. / VRMS).
Thus, the excitation level, depending on the signal between 48 dB and 53 dB, roughly 50 dB below the
Full-range level.
This leaves a dynamic range of
D = - excitation level - noise (dB).
Based on the above-mentioned map means
M-Audio Transit
Realtek
D = - 50 - 120 = 70 dB
D = - 50 - 80 = 30 dB
It can be deduced that sound cards with a noise floor of -80 dB for the measurement with
Noise excitation are practically useless.
Such cards are for measurement, sinusoidal excitation 'but still quite usable (see
Chapter STEPS).
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5
Calibration of the measuring chain
Waiving absolute values of a measurement chain can be operated uncalibrated, but not
without the individual components to adapt to one another. Both a measurement signal of the input
Overmodulated sound card, and one that is hardly perceived, is a reliable
Measurement result is not beneficial.
Therefore, it applies to the construction of a measuring system to analyze the individual components and
possibly by
Coordinated with one another amplifier or voltage divider that neither over-nor a
Can occur under control in a part of the measurement chain.
To set the mood here a first example. The task is a determination of the SPL for
Speaker cone defined in the near field. For this purpose, the electrode to be constructed so that at 130
dB without clipping the input of the sound card occurs. The following values are known:
Maximum input voltage of the sound card UIN MAX = 0.9988 V RMS (see definition below)
GPRE mic gain = 20 dB = 10
SMIC sensitivity = 11mV @ 94db at 1 kHz
At 130 dB - equivalent to 36 dB to 94 dB difference - there is a voltage at the output
Microphone of 10 (36/20) = 63.1 * 11 = 694 mV RMS, which through the mic
is further amplified by a factor of 10.
GIN = UIN MAX / SENSOR MAX
VOUT
= 0.9988 / (10 * 0.694) = .1439 = -16.84 dB
So it is a voltage divider with about 16 to 17 dB attenuation required.
Rx = (ZIN * R2) / (R2 + ZIN)
[1]
G = Rx / (R1 + Rx)
[2]
R1 = (Rx / G) - Rx
[3]
If the input impedance of the sound card of ZIN = 10k and a selected value of
R2 = R1 1 kohm calculated by [1] and [3] as follows:
Rx = (10000 * 1000) / (10000 + 1000) = 909.09 ohms
R1 = (Rx / G) - Rx = (909.09 / 0.1439) - 909.09 = 5408.42 ohms
5.6 kOhm
GIN = 909.09 / (5600 +909.09) = 0.1397 = -17.01 dB
The complete adjustment of the measuring chain is described below step by step.
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5.1
Calibration of the sound card
On the menu SetupCalibrate devices The dialog 'Sound Card and Microphone
Calibration ' opened. The following image shows the preset default values.
Figure 5.1: Calibration Menu
The calibration menu is divided into three sections.
(A) sound card, left channel, output,
(B) sound card, left and right channel input
(C) microphone level calibration.
Note:
Soundcard fullscale input and output are in
Menu Soundcard and Microphone
Calibration ' specified in mV peak.
For the adjustment calculation in the
ARTA MessBox please use mV
RMS = 0.707 mV peak * (see section 3.1)
VS = Vpeak
Vrms = Vrms = 0.707 * VS
VSS = Vpeak Peak
Vmom = instantaneous value
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5.1.1 Calibration of Output Channels
For the calibration of the output channels of the sound card you are working on the following procedure:
1) Connect a suitable
Voltmeter with the left line
Output of the sound card.
Suitable is any AF voltmeter
or digital voltmeter also
at 500 Hz, yet accurately measures, or
an oscilloscope.
The accompanying chart shows the
Deviations of a high
average DMM in
Depending on the frequency.
2) Press the "Generate sine (500Hz) '. ARTA generates a sinusoidal signal with selectable
Maintain amplitude (output level), which is recommended, the default setting (-3dB) ..
3) Enter the data read from the voltmeter / oscilloscope value. The result, in Vs
(Oscilloscope) or be entered in mV RMS (voltmeter).
4) Press' Estimate Max Output mV '
5) The value determined for Max output is in the 'Estimated' displayed ..
6) If the displayed result is plausible, press 'Accept', and the value determined
is the current value for 'lineout Sensitivity ' adopted.
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5.1.2 Calibration of the input channels
For the calibration of the input channels you can use an external signal generator or the
Output channel of the sound card to use as a generator.
If you use the sound card as the example shown here, proceed as follows:
1 Set the volume of the left and right line-in
Channel to maximum (see above)
2 Connect the left channel line-out to the left
Line-In channel.
3 Press' Generate sine (500Hz) '.
4 Enter the value of the generator voltage (in this
Example 676 mV, see above)
5 Press' Estimate MV max input ' and watch
the input level at the peak in the lower part of the display
Mask (see Figure 5.1). If the input channel overrides
is to reduce the input level (see section 4).
6 If the value is plausible, press 'Accept',
and the calculated value is as the current value for 'Line In
Sensitivity ' adopted.
7 Repeat 1-6 for the right line-in channel.
Note: This procedure is recommended as it guarantees that the map in 'loopback
Mode can be used without control '. If the input channels with maximum
Calibrate input volume, then you have the level of the output channel for many sound cards
be reduced by 1-2 dB.
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5.2
Level calibration of the microphone
To calibrate the microphone, you need a level calibrator. The procedure is as follows:
1 Connect the microphone pre-amplifier with line-in
Sound card (left channel).
2 Enter the gain of the preamplifier (preamp gain) and
the SPL value of the calibrator (Pressure) a.
3 Set the calibrator to the microphone
4 Press' Estimate mic sensitivity '.
5 If the measurement is plausible, press 'Accept'.
Note: If the gain of the preamplifier
is unknown, you can set an auxiliary value. This value
but must also as a gain in the 'audio
Device Setup ' be used (see example image 5.3c).
If you do not have a level calibrator, you can apply one of the following methods:
a) Enter the sensitivities of the specification
b) Calculation of the TSP and a near field
c) tweeter as a "reference"
These methods do not replace a level calibrator, however, are for the DIY sector in many cases
sufficiently.
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5.2.1 input of the sensitivities of the specification
If you have a microphone and a microphone with associated and reliable
Data sheets available, enter the appropriate data. Below you will find some
Data for common microphones and microphone capsules. Data for ARTA MessBox see
Chapter 3, the specification of the microphone preamplifier MPA102 in Section 3.1.
Manufacturers and
Description
Thomann T-bone MM1
Superlux ECM999
Behringer ECM 8000
Monacor ECM-40
DBX RTA-M
Beyerdynamic MM1
Audix TM-1
Haun 550 MB
Earthworks M30
NTI M2210
Microtech MK221 & MV203
Sennheiser KE 4-211-2
Panasonic WM 61A
Sensitive
sensitivity
[MV / Pa @
1kHz]
12.9
13.6
12.4
5.6
7
15.2
6.5
6
8
20
50
10
6
Maximum sound- Maximum sound- Dynamics
pressure
pressure
area
[DB]
[DB @ 3%
[DB]
THD]
118
94
129
98
121
91
120
128
140
126
150
145
132
123
103
96
112
142
118
120
about price
€ 35.00
€ 39.00
€ 49.00
€ 84.90
€ 119.00
€ 154.00
€ 295.00
€ 459.00
€ 639.00
€ 1,098.00
€ 1,535.00
146
125
120
For more information on measurement microphones can be found in Chapter 9.2 and
Chapter 1.2.
Image 5.2.1.1: Measurement Microphones (left to right) Haun MB550,
T-bone MM1,
NTI M2210, Audix TM-1
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5.2.2 Determination of the sensitivity of a microphone near field
What to do if a calibrator is available and the sensitivity of the microphone and
Mic is also unknown? Hereinafter, a method is shown with
where you at least get an approximate level calibration.
Take a low-or mid-bass, measure the TSP and build it into a
closed housing a known volume. With the data you enter into
Simulation program and calculate the frequency response of the half space (2 Pi).
Figure 5.2: Determination of Thiele Small parameters with LIMP
If you are still in LIMP (see Chapter 10) have been incorporated, so you use to
First simulation the manufacturer's data. Please use only chassis and data from reputable
Manufacturers, otherwise the calibration could easily lie next few dB.
Figure 5.3: Simulation of a 6 "TMT with AJ-Horn (half space, 2.83 V)
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The picture above shows a AJH simulation example for a 16 cm woofer for a
Input voltage of 2.83 V. The simulated frequency response is to us as objective function (see Section
6.6) are used for comparison with the data of our microphone. Only Prerequisite for
Procedure is that the sound card is calibrated (see Section 5.1).
Please note that the SPL of most microphones / microphone capsules in the DIY
Area is about 120 dB. So start with low levels carefully and avoid
also overdriving the input channels of the sound card.
Assumption: We have no information
on the mic and the
Microphone. Therefore we choose now
arbitrary values, and they give in
Menu, Audio Device Setup ' a:
Gain MVV
Ext left preamp gain = 1
Sensitivity of Microphone
Sensitivity (mV / Pa) = 1
Figure 5.4: Audio Devices Setup
Now we perform a two-channel
By near field and correct
the level on a meter measuring distance.
PFF
PNF = + 20 log (a/2d)
PNF = + 20 log ((12.7 / 2) / 200)
PNF = - 29.97 dB
d = measured distance,
a = radius of the membrane
The measured near field level PNF is
So to correct -29.97 dB to
on the far-field level in a PFF
Meter distance to come.
Figure 5.5 shows the procedure for the determination of the calibration factor from the near field.
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The top image shows the uncorrected
Nahfeldpegel (black line), due to the
lack of calibration course
arbitrary values shows. The red line shows
the imported simulation data.
First you have the uncorrected
Nahfeldpegel means, Edit Scale level '
corrected by the calculated above 29.97 dB
are.
From the remaining difference of the
Calibration factor determined. Shown here in
Example, are about 36 dB difference
recorded.
The corrected again by 29.97 dB
lowered by "scale level" by 36 dB,
the picture is that standing left.
We see that the simulation and measurement now
are largely under cover.
Calculated from this second level correction
the calibration microphone and
Mic as follows:
Gain = 10 (36/20) = 63.0957
Finally, this value is only in
Menu "Setup Audio Devices' in the box
Sensitivity entered.
Attention, any change in the microphone path
(E.g., change of the gain of the
Microphone preamplifier) requires a
Correction of sensitivity.
Figure 5.6: Calibration of a microphone by means of a near-field
measurement
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5.2.3 tweeter as a calibrator
The following "calibration method" relies on the reliability of the manufacturer.
Needed a tweeter as well as the corresponding data sheet. Only use products
reputable manufacturer, because fantasy data are completely out of place here.
Figure 5.7: Data sheet of a known tweeter
The procedure when the calibration procedure is as follows:
1) measuring the impulse response of the tweeter at about 20 cm to 40 cm measured at a distance
small baffle (see Note 1).
Figure 5.8: Impulse response of the tweeter with gate at runtime determination
Correct 2) level measurements at 1 m measuring distance
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For this we first need the actual measured distance. Which must be estimated in two ways:
•
Put a gate: cursor (yellow line)
putting on sample 300 markers (red line)
set to the first spike. The
Result directly in the footer in ms
displayed and is calculated by multiplying
0,344 directly with the measured distance.
d = 0.917 * 0.344 = 0.3154 m
•
Or calculate the measurement distance d as
follows:
d = c * (peak position - 300) / sample rate
= 344 * (344-300) / 48 kHz
= 0.3154 m
Correct the level above in mask Pir scaling 'is shown on a meter measuring distance.
Note: As of Release 1.2, the
Measuring distance by activation of "gate
Time "in the menu" View "directly under the
Displayed graphic
3) Put in the menu overlay '
, Generate Overlay Filter Response '
a target that is approximately the frequency
response
from the one shown in Figure 5.7
Specification maps.
This are various filtering options
first to sixth order to
Available (see right). Filter type,
Sensitivity and corner frequency are
freely determined.
Figure 5.9 shows the measured and
one meter corrected frequency response
together with the target feature (12 dB
Butterworth, fc = 900 Hz).
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Figure 5.9: Measured frequency response and target
4) Calculate correction factor
From the frequency response, we can set the cursor for frequencies at least one
Octave above the resonance frequency, reading the corresponding level values. Analogous to the
preceding example, the correction factors are calculated now.
KE 4-211-2
Simulation
ARTA
Difference = - (SPL simulation - SPL
measurement)
10 ^ (Differenz/20)
adopted amplification
setting of amplification
3000 Hz
92.00
104.49
12.49
4.2121
1
4.2121
4000 Hz
92.00
102.94
10.94
3.5237
1
3.5237
5000 Hz
92.00
102.99
10.99
3.5441
1
3.5441
6000 Hz
92.00
103.08
11.08
3.5810
1
3.5810
7000 Hz
92.00
103.51
11.51
3.7627
1
3.7627
Thus, an average correction value of 3.7247 with a standard deviation of results
0.2884.
Note 1: Note that when this method is that the size and shape of the baffle,
as has also the location of the tweeter in the baffle effect on the frequency response. The
Impact of the installation conditions can for example be simulated quite true with EDGE
(See Figure 5.10).
Figure 5.10: Simulated influence (red) of a 25x25cm baffle at 30cm.
Ideally, you should choose the size of the baffle so that their influence in the frequency range of
Calibration is as low as possible (see also IEC baffle in Section 11).
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5.3
Compensate for frequency response errors of the microphone
Basically, the use of a good measurement microphone with a linear frequency response
recommended. Suitable specimens that are still affordable for the DIY area, see the
Section 5.3. When you purchase the microphone or the microphone capsule also that it
besides having a smooth frequency response and omnidirectional polar pattern.
ARTA and STEPS offer the opportunity to correct the frequency response of your microphone. It
should be emphasized that this correction is limited exclusively to a measurement axis (in the
Rule 0 degrees). Frequency response measurement error outside this axis are not in the correction
considered.
In the menu, Frequency response compensation ' All the necessary steps to correct the
Microphone included. Proceed as follows:
a) Download the correct file. MIS with, Load ' (See Figure 5-11).
In the compensation file is a normal ASCII file of . Txt in . Mic
has been renamed. The structure of the file must be as follows:
Frequency (Hz) Magnitude (dB)
0.9917.527
0.9517.714
0.9117.902
0.8718.093
0.8318.286
So you can for example read the values from the frequency Wrote your microphone and a
Enter ASCII file without formatting. Make sure that the comma as a point
is entered, otherwise you get an error message.
After the file is loaded, the frequency response of the microphone as in the above example is
displayed. It is important that you enter the frequency response and not already the correction values
(Mirrored frequency response) of the microphone.
If you had only a few measured values (points) are available, produced the ARTA
Intermediate values automatically by a cubic spline. Note, however, that at least
one value per octave should be available and these values evenly as possible over the
Are distributed correction range.
b) Activate the compensation by, use frequency response compensation '(Figure 5.11),
You can in the main menu of ARTA, see Setup, verify that the
Compensation microphone is active.
If, FR Compensation ' is provided with a hook, the
Compensation active. Clicking again will the compensation
disabled.
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Figure 5.11: Menu frequency response
compensation
The above procedure for the correction of your microphone requires first that the
know individual frequency response of your microphone. To the frequency response of the microphone
reach, there are basically the following:
-
Make use of the frequency supplied by the manufacturer Wrote (usually only one
Type specification),
-
Contact a "calibration" (eg hi-fi DIY, IBF Acoustic)
-
Perform the calibration itself, provided they have access to a high quality
Measurement microphone have:
> Substitution method for f> 200 Hz,
> Pressure chamber method for f <200 Hz
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5.3.1 Calibration using a reference-quality microphone> 200 Hz
If you temporarily have access to a high-quality measurement microphone (see eg Picture 5.12a), can
You calibrate your microphone itself.
A pretty good description of the procedure is as on the homepage of Earthworks in the
Find article "How Earthworks Microphones Measures". Earthworks used in
above 500 Hz, the substitution method in which the test object on a infinite
Baffle is measured against a reference microphone. The deeper the test frequency, the
problematic it is to find a suitably large and anechoic chamber or
the measurement of outside influences kept free. To rid yourself of these constraints,
Earthworks used in the lower frequency range, a small pressure chamber for calibration (see
Section 5.4.2).
Image 5.12a: "reference microphone" MK 221 of Microtech Gefell
Image 5.12b shows the reference microphone and the calibration object (MB550) measured
Frequency responses. Calibration object and reference differ in the level and frequency response.
First, we try to compensate for the difference in level, because that would only later than
Make noticeable offset.
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Image 5.12b: reference microphone (MK 221, blue) and calibration object (MB550)
With, Scale Level ' we reduce the level of 550 MB
extent, to the largest possible part of both frequency responses
comes in coverage (see Figure 5.13).
For the best value is not always at first sight
seen, therefore, is useful to try a little. You can
with, often add scale level 'any values or
subtract.
Figure 5.13: Scaling and subtraction
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Then we train with, Subtract overlay ' (See Edit menu in Figure 5.13), the difference between
the two frequency characteristics.
Figure 5.14 shows the result of this operation. The deviations in the frequency range of 150 Hz
to 20 KHz in a band of max. + / - 1.25 dB.
Figure 5.14: Variation of the frequency response of the reference axis
Through export ASCII ' We can now create our compensation file. After renaming
to by *. txt *. MIC can be read as shown above.
5.3.2 Calibration below 500 Hz in a pressure chamber
As already stated, Earthworks used in the frequency range below 500 Hz, a
Pressure chamber for calibration. Construction and operation of the pressure chamber are
ARTA Application Note No. 5 described in detail [V].
Is the largest dimension of the chamber up to 1/6 to 1/8 of the wavelength of the upper
Are cut-off frequency at 500 Hz that is 11.5 cm to 8.4 cm.
Figure 5.15: Design and application of the principle of measuring chamber
The application of the pressure chamber is largely to refer to Figure 5.15. To be tested
Microphone is introduced with the help of an adapter in the chamber, well sealed with putty and
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then measured with ARTA and STEPS in the valid frequency range. The introduction of the
Microphone in the chamber has the advantage that the measurement is largely independent of the
Environment and additional faults are hidden. As usual in the chamber at
Voltage values with very high sound pressure levels (eg 2.83 V 145 dB) is expected, should the
Avoid damage to the microphones to be examined only very small
Used excitation voltages (approx. 0.01 V). Figure 5.16 shows the calculated with STEPS
MK 221 in frequency response of the measuring chamber.
Below is shown by an example, as of the reference curve and the measured curve
for the microphone to be calibrated, the calibration curve is determined.
Figure 5.16: 550 MB (black) and reference frequency response for level adjustment
In general, it is assumed that the microphones have different sensitivity
are. Therefore, first a level adjustment is required. The easiest way is when a
Chosen reference frequency and the corresponding sensitivity is read from the cursor. The
Difference is then purified, scale balance '.
Provided with ARTA was measured, the required difference can be directly connected, Edit
Subtract overlay ' occur. When was measured with STEPS (better reproducibility), is
a small detour through example Excel, use a suitable simulation program (eg
CALSOD) are required.
Figure 5.17 shows the result obtained with STEPS for the microphone 550 MB for the
Frequency range of 5 to 500 Hz Will this compensation curve with the result from the
previous section together (see Figure 5.14), so you get a
Compensation file for the entire frequency response of about 5 Hz to 20 kHz, such as in Figure 5.11
shown.
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1.0
Deviation 550 MB
from the
0.0 in
reference
dB
-1.0
-2.0
-3.0
Reference microphone Microtech Gefell MK 221
-4.0
1
10
100
1000
Frequency in Hz
Figure 5.17: 550 MB, deviations from the reference frequency
response
The results of additional microphones are summarized in Figure 5.18. They show that below
calculated from 100 Hz with significant variation between different DIY microphones
must be. Even relatively high-quality microphone capsules (211-KE 4) are apparently not
Guarantee that deviations from the specifications or deviations are negligible.
Figure 5.18: Results of the tested microphones: Black (MB 550), Red (211-KE 4, No.1)
Light Blue (211-KE 4, No.2, Nr2K), Blue (MCE 2000), Orange (Panasonic WM 60)
That are adjacent to a flat response, other criteria of importance to by
Figure 5.19 are shown. All microphones were already mentioned in the pressure chamber
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Microphones in a small pressure chamber @ 300Hz
10.00
1.00
THD in%
0.10
THD (%)-MK221
THD (%) MB550
THD (%)-KE 4-211
THD (%) MCE2000
THD (%)-WM60
0.01
110
115
120
125
130
135
140
145
Inside box in dB SPL
Figure 5.19: Comparison of the harmonic distortion of microphones at 300
Hz
studied in terms of distortion behavior. This clearly shows in distortion (THD)
and SPL, why professional measurement microphones "something" are more expensive.
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5.4
Testing of the measuring amplifier
An essential part of the measurement chain is the measuring amplifier. In many cases, the separate
When measuring power output stage are used occasionally probably also kits or
House developments. Regardless of the choice, it is desirable, at least the basic data of the
amplifier used to know.
If the amplifier is used only for usual frequency response and impedance measurements,
are linear amplifier with a frequency response between 10 Hz and 20 kHz and services
Range of 6-10 watts sufficiently. If, in addition to distortion and power compression
Be measured speakers outputs of 100 watts at 8 ohms and more are not harmful
Lich. Order now own shank device relating to these conditions under the microscope to take
can, here's a little digression for measuring amplifiers with ARTA.
For the measurements with ARTA we use the test setup of Figure 5.20. In this way we
Ensure that the input channel of the sound card is not overloaded or when surge through
the diode is protected.
A = 20 * log (Rx / R2 + Rx)
Rx = ZIN * R1 / (R1 + Zin)
Example:
ZIN = input impedance of the sound card
= 10k
Weakening AR1R2
-10 DB510 Ω 1047Ω
-20 DB510 Ω 4.4 kΩ
-30 DB510 Ω 15kΩ
Figure 5.20: Voltage divider for ARTA for measuring amplifiers
As an example, the measurement was " t.amp " PM40C selected from Thomann. The manufacturer
published
following specification.
Technical Specifications
Output Power
into 8 Ohms: 36W rms
into 4 Ohms: 50W rms
Frequency Response: 10Hz - 20kHz / - 1dB
Voltage gain: 26 dB
Input Impedance (balanced active): 20 kOhm
THD + N: 0.03%
Slew Rate: 19 V / ĩS
Signal-to-Noise Ratio: 92 dB
Power Consumption: 75 VA max.
Dimensions (WxHxD): 155 x 166 x 55.5
Weight: 1.8 kg
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Figure 5.21 shows the harmonic distortion of the t.amp to 4.1 ohms (black) and 8.2 ohms (red)
Depending on the output voltage. The t.amp are about 34.0 watts into 4 ohms and about 23.2 W
undistorted at 8 ohms again. For measurement purposes, so should not the t.amp by a stop
be, with 10 volts rms output voltage to 4.1 ohms (24 W) is located on the safe side.
Figure 5.21: THD @ 1kHz as a function of the output voltage at 4 and 8 ohm load
The measurement of the frequency response is shown in Figure 5.22. Thereafter, the lower
Cut-off frequency (-3dB), about 16 Hz and the upper limit frequency of about 60 kHz.
Figure 5.22: Frequency response of "T.AMP" from Thomann (based LM3886)
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Figure 5.23: THD + N @ 1kHz and 1dB
Figure 5.23 shows the measurement of THD + N for the t.amp. The values obtained are within the
Manufacturer's instructions. Figure 5.24 and Figure 5.25 show the Klirrfrequenzgang the t.amp at 1 and
16
Watts into 8 ohms. Up to 16 watts, the t.amp shows obviously unimpressed.
Figure 5.24: Klirrfrequenzgang the "T.AMP" at 1 watt into 8 ohms
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Figure 5.25: Klirrfrequenzgang the "T.AMP" at 16 watts into 8 ohms
As of release 1.3 are shown in STEPS stress-and performance-related distortion measurements
possible. Figure 5.26 shows the voltage-dependent harmonic distortion of the amplifier for three
different frequencies. More details on this measurement, see Section 9.3.
Figure 5.26: Voltage-dependent harmonic distortion (THD) of the measuring
amplifier
to 4.1 ohms at 100 Hz, 1 kHz and 10
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Besides power, frequency, phase response and harmonic distortion of an amplifier in addition, the
Parameters from Figure 5.27 of interest.
Source
Input voltage
UE
Impedance
RS
Amplifier
Load
Voltage gain
V = UA / UE
Input resistance
RE
RE RS <<
Output voltage
UA
Output resistance
RA
RA RL <<
Load resistance
RL
Figure 5.27: Schematic diagram of an amplifier
The Input resistance RE is the input-side impedance and an amplifier
determined by the fact that the amplifier input resistor connected in series RV.
Characterized the input voltage goes from UE1 to UE2 and with it, the output voltage of
Back to UA1 UA2. This results in the input resistance of the amplifier to:
RE = RV * UA2 / (UA1 - UA2)
T.AMP example: RV = 47 kΩ;
UA1 = 10,502 V; UA2 = 3.144 V
RE = 47 kΩ * 3,144 V /
(10,502 V - 3,144 V)
= 20.082 kΩ
Figure 5.28: Measurement of input resistance
The Gain is the ratio between the output and the input voltage
an amplifier.
V = UA / UE
It is measured with a sinusoidal alternating voltage whose frequency is typically 1kHz
is. A precise voltage divider between the generator and the amplifier facilitates
Measurement at high amplification factors (eg microphone). Measuring the voltage
UE 'in front of the voltage divider and multiplied with v u = the voltage divider ratio (R1 + R2)
/ R2. Then V = U * UA / UE '.
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V = UA / UE
T.AMP example: UE = 0.8493 V;
UA = 18.539 V
V = UA / UE = 18.539 / 0.8493
V = 21.83 = 26.7 dB
Figure 5.29: measuring the gain
The Output resistance is the internal resistance of the output side of an amplifier and
determined by the fact that one loads the output with a resistor RL. This reduces the
Output voltage of the open circuit voltage U0 to the value of the terminal voltage UL
from. The output resistance is then
RA = RL * (U0 / UL - 1).
T.AMP example: U0 = 5.470 V;
UL = 5.462 V, RL = 8.2 Ω
RA = 8.2 * (5.470 / 5,462-1)
= 0.0120 Ω
Figure 5.30: Measurement of the output resistance
Are the measured values for the input and output impedance and the gain of
They each next to the pictures 5:28 to 5:30. The measured values for RE and V are in
Within the manufacturer's specification.
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6
Measured with ARTA
6.0
General
After the calibration of the measuring chain is complete and everything prepared for measurement, can
start it with the actual measurements. One should make it suitable for iron rule, before
each measurement session, all cable connections and settings thoroughly and calmly to
control.
Image 6.0.1: Measuring equipment without measuring or connecting cable and
tripod
Cheap cable - and here especially bad compressed cable with alligator clips - or
Fast soldered together the connecting cable solve many mistakes, long viewfinder land
and often despair.
NOTE: A well put together with high quality measurement equipment and
clearly marked measuring or connecting cables and a ARTA MessBox
help nonsense errors (and damage) to avoid. This is especially true when
a long time was not measured and the familiarity with the system lost a little
has come.
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6.0.1 Test Leads
Who wants to measure small analog voltages should also pay attention to its measurement cable. Naturally
suffers the transmission quality of the signal when measuring small signals with simple cables,
larger distances are transmitted, as noise of any kind act on the pipes.
To counteract the mass and interference problems, the following guidelines [26] should be considered
are:
•
•
•
•
•
•
•
•
Use the shortest possible cable between the source (sensor) and measuring amplifier.
Especially in the case of high-impedance sources has to be ensured.
If possible, use double shielded cables.
If necessary, drag an extra ground wire and connect the shielding
on one side only.
Avoid ground loops. Pay attention to the same earth potential between the measuring source
and meter (sound card). Measure before with a DVM between two
Ground potentials in AC and DC.
Do not place the signal cable to pass interference (transformers, power supplies, power supply
leading cable, etc.).
If possible, disconnect the computer from the network electrically (laptop battery)
Take advantage of the additional averaging (averaging)
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6.0.2 The signal / noise ratio of the measurement chain
The observance of Signal-/Störabstandes (S / N ratio) is used for each measurement, a special
Importance. A correct frequency and phase for a measurement can only be calculated
be if the signal or useful level is greater than the noise level.
Therefore, should be determined before each measurement session, the signal /
noise ratio.
Measure to the intended measurement setup the sound with and without speakers (DUT)
and compare the levels (see Fig .6.0.2). The noise should be in the interest
Area of at least 20 dB below the signal level. Where: The greater the distance, the
better the quality of the measurement results.
Image 6.0.2: Determination of Signal-/Störabstandes
Shows no or only a little in one or more relevant for the measurement bands
Level difference, you have the following options to improve the situation:
-
reduce the noise level or change the room or the measurement environment
increase the level of the excitation signal
Avoid excitation signals with low energy content (eg MLS)
Averaging, see section 6.0.3
The phase transition is very sensitive to an unfavorable signal / Störabstandsverhältnis and
this especially for measurements on loudspeakers and speakers that are not the entire
Frequency range covered. In principle, the phase frequency response only there reliable
can be calculated, where a sufficiently large S / N ratio is found.
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Image 6.0.3: Frequency and phase response of a HT measured in normal housing
For measurements on each of the speakers, this is not usually the entire frequency response
the case. So shines a tweeter at 100 Hz as little sound energy from that
Transfer function in this frequency range covered by the noise floor of the measuring room.
The phase characteristic is therefore largely calculated noise from the space, and is therefore
not usable.
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6.0.3 Averaging
As mentioned above, measurements are rarely held under optimal conditions. Often has
is considerable noise from traffic noise, fan noise from computers, the
Start of heating or air conditioning systems, as well as wind noise in the building.
To this measurement results with tolerable accuracy
get it relies on the averaging. In Fashion IMP
we find in the "impulses response measurement " the
"Number of averages ". In Fashion FR1, FR2 and SPA
see the submenus under "averaging"
the "Max field averages ".
Image 6.0.4: Averaging in IMP mode
These fields specify the number of measurements is set ARTA then forms
automatically the average of these measurements.
Per doubling of the number of measurements, the signal to noise ratio is increased by 1 / √ n, that is 3 dB.
However, this can not continue as desired, since other phenomena, such as Jitter,
the whole sets a limit.
6.0.4 Image shows the measurement result for the noise level of 2, 4, 8, 16 and 32
Averages. As we see, the averaging method is quite effective.
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6.0.4 ARTA excitation signals
ARTA provides a wide range of tightly integrated excitation signals, and additionally the
Opportunity to work with external stimulation. To below the respective menus and
their signal range.
Impulse Response
Periodic noise (PN: Pink,
White, speech); Sweep: Sine (lin,
log); MLS; External stimulation
(Trigger)
FR2, FR1, SPA
Random (Random): Pink, White
Periodic: White, Pink, Speech
The difference between
periodic and random noise
is exemplified in Figure 6.0.6. illustrated.
Starting with version 1.6.2
Signal Time Record
Signal generator with a large
Range of continuous signals
(Sine, square, multi-tone, etc.) and
Pulse and burst signals (see also
Picture 6.0.8)
The selection of an optimum excitation signal depends on both the quality of the used
Hardware (sound card) as well as of the respective measurement environment.
Dual-channel measurements should be used with high-quality sound cards over a wide,
linear frequency response and have close tolerance sensitivities of the input channels,
Single channel measurements can, however, also with lower quality sound cards
be carried out (see also Chapter 4, last paragraph).
For the selection of an appropriate excitation signal are by Ivo Mateljan following instructions
given:
•
•
In an environment with high noise are periodic noise (PN) the best
Results. Averaging always improves the S / N ratio. It reduces the impact of
random and stationary noise and nonlinear distortions.
In quiet surroundings of the high crest factor makes the sine sweep is ideal for highPower-speaker tests. When using sinusoidal sweeps the averaging does not bring
always improve the S / N ratio. Here it is better for the duration of the sweep
increase.
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For periodic noise (pink) ARTA provides for
Protection of the DUT against low-frequency, high-energy
Signal components, a kind of high-pass filter on (Pink cutoff).
The effect of Pink cutoff Image is 6.0.5
described. Will increase with the cutoff frequency
increasingly a low-frequency cutoff level
made. This level capping is ARTA
automatically compensated for mathematically.
Image 6.0.5: Effect of "Pink cutoff" with 10, 20, 50 and 500 Hz
In the following still "shots" shown by other signals (Figure 6.0.6 to Figure 6.0.7).
In-depth comments on the topic, please see the original manual or also [30].
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White
Noise
Pink Noise
Random
Periodic
Image 6.0.6: Difference between random (random) and periodic noise
Image 6.0.7: Multi-Sine
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As of version 1.6.2 ARTA contains an additional signal generator with the continuous
Signals (sine, square, multi-sine, etc.), pulses (eg Dirac) and sine burst of various kinds
can be produced. The application of the sine burst the side of Siegfried Linkwitz is
recommended (http://www.linkwitzlab.com for example, "Triggered burst measurements of tweeters").
Picture 6.0.8 shows the menu "signal generation and recording." The particular choice of the waveform
done by clicking on the checkbox "Continuous", "Pulse" or "Sine burst". After this election
the signal in each case to be more specific (eg, type, frequency) and at the transient
Adjust the frequency of repetition (repetition). So does a high 16,384
Repetition, while 262,144 - depending on the choice of sample rate and Lenght field signal
Recording - possibly contains only one repeat per record.
With the checkbox "Invert output signal", the output signal is inverted, with "trigger on right
channel "can be for two-channel measurement setup the recording by the output signal of the
Sound card control.
Image 6.0.8: menu "Signal generation and recording"
The checkbox "link" between the buttons "Generate" and "Record" automates the
Trigger process, both processes are coupled.
The two fields "signal recording" and "triggers" are self-explanatory or from
other contexts already known.
Picture 6.0.9 shows a collection of signals from the "Transient Generator". In the left
Half of the picture you can see the excitation signals and the right half of the image with a
high quality microphone recorded response to a tweeter at 3 kHz.
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Triangle
Window
Gaussian
window
Uniform
Window
Pulses,
width = 20
Image 6.0.9: And burst pulses: excitation (left), response (right)
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6.1
Impulse responses - Theory and Practice
Depending on the measurement object - especially with subwoofers - and knowledge of signal theory, the
More or less great surprise when the first impulse response can be seen on the monitor.
Therefore, at this point, a brief overview is provided with examples from theory and practice
are.
Image 6.1.2.1: Step response (middle) and frequency response (right) of a Dirac pulse (left)
To describe the theory of a Dirac pulse was (see Figure 6.1.2.1) with respect to the target
(Low-pass, band-pass, high-pass) filtered and then loaded as wav file in ARTA and
evaluated. In this way generated impulse and step responses and frequency responses
match - if the bandwidth limit does not thwart the bill - the
ideal curves of the filter theory
Image 6.1.2.2: Impulse response (left), step response (middle) and frequency response
(right)
a 1000 Hz low-pass filter
Image shows 6.1.2.2 as the first example, a 12 dB low pass filter with a cutoff frequency of 1000 Hz
Note the changes in impulse and step response in comparison to the image 6.1.2.1.
Image 6.1.2.3: Impulse response (left) step response (middle) and frequency response (right) of
Band-pass filter with 100 Hz / 1000 Hz Crossover frequency
The second example in Figure 6.1.2.3 shows a 12 dB band pass at 100 Hz 1000 Hz respectively
Cutoff frequency. Also analyze here the changes in impulse and step response in
Compared to the Dirac impulse. Please note the different timelines.
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To the effect of different cut-off frequencies of the appearance of the step response
illustrate in the next picture ever a 12dB low pass, band pass and high pass is shown.
Do you have a particular attention to the band-pass, because all speakers show this
Behavior.
12dB
lowpass
500Hz
1000Hz
5000Hz
30 - 100Hz
100 - 1000Hz
3000 - 22000Hz
20Hz
50Hz
1000Hz
Bandpass
12dB
High pass
12dB
Image 6.1.2.4: Influence the cutoff frequency of the appearance of the step
response
The last example is a tweeter with a 12 dB high-pass corner frequency of 1000 Hz
simulated. For this purpose, there is a real counterpart. Image 6.1.2.5 shows the simulation (top) and the
measured frequency response of a tweeter. As we see, there are significant differences for
theoretical course. Both the individual characteristics of the tweeter, and the
Show installation conditions and the conditions of measurement (measuring distance, space, noise, etc.)
in the impulse response and, consequently, in the secondary analysis. To the strange
During the phase response is an explanation in chapter 6.1.
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Image 6.1.2.5: Impulse response and frequency response of a simulated and a real tweeter
In addition, of course, is also a tweeter through a bandpass to describe, but here
due to the bandwidth limitation of the simulation software (22 kHz) and the measurement system (24
kHz) restricted displayed (Figure 6.1.2.6)
Image 6.1.2.6: Simulation as a tweeter high-pass (left) and band pass (right)
Finally, a note on a frequently asked question: Where do the
strange artifacts before the actual impulse response?
Image 6.1.2.7: Impulse response using pre-ringing
This so-called pre-ringing is a result of the bandwidth limit of the measuring system. It
So each occurs at frequencies half the sampling rate, in today's conventional sound cards
24 kHz (48 kHz) and 48 kHz (96 kHz). Limited Remedy can dual by setting "filter
channel impulse response will be reached "in the menu" Impulse response measurement ".
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6.2
Measure, but where?
Before answering the question of where, first is the question of what to
what context. The "what" is defined in this context an important part of
Measurement task. If for example a subwoofer or a 3 way floorstanding speaker to be measured, are
to fulfill other conditions as if a small speaker for the desk - equipped
with a small full range - will front the microphone.
Image 6.2.1: Simulation, 3-way crossover
To give an example: Image Image 6.2.1 and 6.2.2 show two fictional Lautsprecherkonstruktions. For the development of the 3-way crossover box measuring 2 Ok should also
octaves below the transition frequency of DD / MT - 300 Hz in the example - enough resoluhave the solution and at a measurement distance including the integration of the two speakers
effects of housing permits (see section 6.3). A beautiful illustration of possible speakers and
Housing effects to be taken into account when interpreting the results of the measurement and,
is followed by J. Backman image to see [29].
Speaker and enclosure effects [29]
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Even a 2 way speaker with a crossover frequency of about 2000 Hz lower than required
Frequency limit of at least 500Hz (Figure 6.2.2). If the so-called Baffle Step (See right
Partial image) can be taken into account in the development of course, it must be - depending on
the
Baffle width - reflect the measurement 200-150 Hz with sufficient resolution.
Image 6.2.2: Simulation, 2-way crossover (left), TT with / without baffle step (right)
There is still the requirement that all possible repercussions such as room reflections or
standing waves of measurement are to be kept, so the task is not just
easier.
Before going into the details, let's look first look at what kind of solution
Measuring tasks in the "tool box" is. In the literature [2, 22-26], the following options are
discussed: free-field measurement, measurement in an anechoic chamber, ground-plane measurement
Half-space and windowed measurement and field measurement (Section 6.2.0).
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FREE FIELD MEASUREMENT
As accurately described by the name, the first and oldest way is the measurement in
the wild. The speaker and microphone are reflecting the extent of all
Appropriate areas that are practical no effect of reflective surfaces - in the
Usually the ground - are more. For this, a crane, a tower or similar is required. Picture 6.2.3
right part of figure shows an example of the practical measurement setup [22]. The speaker and the
Microphone can be conveyed by means of a "lift" to a lattice mast at the measurement position.
The reflected sound scores from ((2 · H + d) / 344) seconds at the microphone. The left part of image
shows a simulation in which the ground reflection the direct sound at 1, 2, 4 and 10 meters
is superimposed. It is clear that in reverberant floor (worst case) for a reasonably
interference-free measurement of the height of the tower should be in the range of 10 meters.
Image 6.2.3: Free-field measurement, simulation, ground clearance (left), measurement setup [from
22] (right)
Besides the advantage of being able to create theoretically ideal measurement conditions, an aspect is in
the
Free-field measurement of course always be observed: The weather! Not only snow and rain, and the
Wind and noise make life difficult and thus measurements only
preferred climates reasonably predictable.
Nevertheless, one who calls his own quiet garden, the free-field measurement should not leave the
Lose sight of. Even if your own "pylon" is only 3 or 4 feet high, in conjunction with
a "windowed" measurement are now 40 - 50 Hz lower cut-off frequency as feasible (see
Section 6.2).
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POOR REFLECTION ROOM
If free-field measurements are to be carried out regardless of the weather and background noise, then
only helps a anechoic chamber (RAR), sometimes anechoic or echo-free space
mentioned. In a RAR are all partitions with
Sound absorbing material - usually glass or
Mineral wool - lined. To the fullest possible
To achieve sound absorption in the entire useful range,
the lining is often performed in a wedge shape (see Figure
6.2.4).
RAR can run as a full space or half space
are. In a room full all interfaces with
absorbent material provided. The accessibility of the
Space is collected by a grid floor or a
Span wire mesh guarantees (see picture left). In a
Half of the floor space is reverberant and is therefore without
Restrictions available.
Image 6.2.4: RAR Visaton [24]
RAR quality are "room within a room" construction.
The usable space inside is completely contained by the springs
other building decoupled. Thanks to this design
the transmission of airborne and impact sound strong
reduced, a low noise level
guaranteed
By the absence of reflections, the sound field of a RAR corresponds to the outdoors in a large
Distance above the ground (see also free-field measurement). The light emitted from a sound source signal
unaffected by the room.
Image 6.2.5: Relationship between frequency and wavelength
The lower frequency limit of a RAR is determined by the dimensions of space and
Lining determined. Conventional cut-off frequencies in the range of 70 Hz - 125 Hz, and
put a gross volume of 350 m3 to 60 m3 ahead. The absorption length of the wedges
should be about 1/4 of the wavelength of the lower frequency limit amount (see Figure 6.2.5). To the
to above normal cutoff frequencies still effectively absorb, are thus
Wedge lengths of about one meter is required.
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GROUND PLANE MEASURE
An interesting aspect of the ground plane measurement - hereafter called short-GPM is that
neither space nor tower, but a large reflecting surface is necessary. One
paved parking, a playground or a great gym are - of course outside
the normal period of use - suitable objects.
Image 6.2.6: Ground-plane measurement
There should be no reflecting obstacles in the vicinity of the measurement site. The distance from
the source (speaker) to the next obstacle should be at least five times the
Be measuring distance. This ensures that the level of the reflection by at least 20 dB
is reduced and less than 1 dB contributes to the overall sound pressure.
The speaker should be on the ground and be tilted so that the loudspeaker axis
directly points to the measurement microphone. The microphone must be located directly on the floor
(Figure 6.2.6).
The angle α is calculated as follows:
α= Arctan (H / d)
H= Distance from bottom - middle of the
membrane
d= Distance Microphone - Speaker
The measuring distance shall be large enough to safely be in the far field. In general, the
assured if the distance is greater than three times the maximum dimension of the
Source, this source and mirror source must be included. Must in principle
GPM be noted that two sources are mirrored along the measuring axis. The
Baffle is therefore twice as large, and the shape is different from that of a single system.
In the case GPM effects should therefore always be carefully considered. Since this is the
Has significant effects in the vertical direction can Polar measurements or
Distortion measurements are performed as usual.
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Image 6.2.7: GPM, free-field and half-space measurement at 1 meter
For the measurement of small loudspeakers or loudspeaker chassis has in practice
Measurement distance of 1 meter by set as default. It should be noted that the GPM
Mirroring the source level to the axis 6 dB adds. It may therefore be desirable in
GPM to increase the measurement distance of 2 meters, as reduced by the doubling of distance
the level by 6 dB.
Is encouraged provided with the same input power, has a GPM at 2 meters distance measurement
the same sensitivity in the mid and high frequencies as a 2π or 4π measurement at 1
Meters. At low frequencies, which level is identical to a 4π measurement. Then follows a
Region in which the radiation source - as a function of the size of the
Baffle and its mirror image - slowly changed from 4π to 2π.
HALF-SPACE
For a half-space measurement (2π) either the floor or a wall serves as a "infinite
Baffle "for the speaker to be measured. In open terrain, a pit excavated
are in the building, the flush sinking of the speaker
the floor or the wall, a not insignificant structural
Constitute a measure (see eg www.hobbyhifi.de, measuring room).
For measurements in nature are essentially the statements
the free field (cf. above picture and 6.2.7). With
Measurements in space are the versions from 6.2.0 to Chapter
note.
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6.2.0 Measure under housing conditions
Usually not a DIY controller via a gymnasium, a 10 meter high tower, or even a measurement has
anechoic chamber. One must, willy-nilly with living rooms and basements - or
in the summer and with no wind with gardens and parking areas - content.
What is observed in measurements in "confined spaces" and how can it ARTA
support? How do the measuring chambers of professionals from normal living rooms?
The first answer to this question in Figure 6.2.2 are two different measurement spaces
compared. The test object and the measurement conditions were as part of a collaborative study
(Http://www.visaton.de/vb/, keyword proficiency test) is clearly defined, and the two measurements
constant. The only difference between the conditions was measured in the measuring chamber,
documented by the bottom row in Figure 6.2.2 using the reverberation time courses. During
the anechoic chamber measuring well below RT = 0.15 s, the housing moves in
Medium at RT = 0.35 s
The measuring distance was 30 cm, the test object, an 8cm full-range loudspeaker Visaton, was
flush mounted on a small baffle. The chassis and the measurement microphone were located in
approximately half the room height.
In the upper row, the unsmoothed frequency response is seen. Clearly make the
Reflections of the living space in the right frequency Wrote noticeable. The second image set of
above shows the smoothed with 1/24 octave frequency responses (black curve). As clearly
is seen, the roughness of the frequency response is maintained. Only by setting a window (see
Picture 6.2.1), the room reflections disappear (red curve).
Image 6.2.1: Hiding the room reflections by setting a window
Even in the period-based Waterfall (third row from the top picture) is the living space to clearly
. seen The "roughness" in the frequency response between 200 and 2000 Hz are manifested here
slower decay of the vacuum-energy noticeable. To illustrate this, this process is also
normal waterfall diagram (fourth row from the top picture) are visible. Interim conclusion: The Professionals
have it easier to separate the speaker from the room during measurement.
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Image 6.2.2a: Comparison of two measurement chambers (see Figure
6.2.2b)
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Image 6.2.2a: characterization of a measuring chamber (L = 4.95, W = 3.85, H = 2.25, RT =
0.38 s)
Area I
Pressure range
f = c / (2 · L) where c = 344 m / s, L = length of
chamber
Region II
Resonance range (room modes)
f <= 2000 · √ V / RT = room volume V, RT = reverberation time
Area III
Statistical area (diffuse or reverberant field)
f> = 2000 · √ V / RT = room volume V, RT = reverberation time
Nevertheless, it goes without anechoic chamber. Struck and Temme [3] describe how
you can "simulate" free field measurements in normal rooms. To be a
Near-field and far-field assembled. The definition of local,
Remote and free field can be derived from Figure 6.2.3 quite well.
Near-and far-field refers to the distance from the sound source, free (or direct) and
Diffuse-field contrast to the environmental conditions of the sound source.
The Free and the diffuse field are independent of the type of the sound source, they are the
acoustic properties of the room environment, characterized in the sound source. Spreads the sound
in all directions from the sound source without hindrance, ie there is noise in the observed field
no obstacles reflected or scattered sound waves, one speaks of
Field conditions.
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Free field Only direct sound reflections without
Near field Measuring distance <emitted wavelength
Far-field Radiated wavelength> source dimension
Sound pressure decreases with 6 dB per doubling of distance from
Image 6.2.3: Definition of sound fields
The sound source is located in a room, the sound waves are radiated
Room surfaces or furnishings reflected. The multiple reflections
is a complete mixing of the sound waves, ie at each point in space is the
Incident sound from any direction in space are equally likely. The local sound energy density is at
all points in this field mixing equal, if the microphone is sufficiently far
is away from the sound source and from all reflective surfaces. One then speaks of the diffuse
Sound field.
At the sound radiation in a room predominates in source close to the free field, in a
sufficiently large distance from the source, the diffuse field. As a boundary between these two
Image 6.2.4: Definition of Sound Fields, Hall radius
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The distance from the sound source is defined field types, wherein the components of the two sound
fields
are equal in size, the so-called reverberation radius RH
RH = 0.057 √ (V/RT60) V = volume [m ^ 3] and RT60 = reverberation time [sec]
If the distance from the sound source is less than the critical distance, then the sound field in the room is
in the free field on the source.
Image 6.2.5: Determining the reverberation
radius
Example: For a room with a volume of 50 m ^ 3 (5 x 4 x 2.5 m) and a reverberation time
of 0.4 sec, the reverberation radius is about 0.64 meters. If the measurement distance of 1 meter in safe
Free field are, as in a room of the same size would be a reverberation time of 0.2 sec well below
to implement.
Note: To measure the reverberation time see Section 6.2.1.
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What can we do with this information? Now, from this we can deduce when we
move us in our room measuring predominantly in the open and they allow us a rough
Assessment of the expected measurement quality.
Image 6.2.6: Positioning the microphone in the short-, long box *) see note 1
Information for near field:
•
•
•
Microphone as close as possible and centered on membrane
Measuring distance <0.11 * dimension of the source error <1 dB
upper frequency limit for near-field measurements obtained from image 6.2.7
Regarding near-field measurements must be considered two things. For one, that the microphone
is not overloaded and secondly, that the scope of near-field measurements to
higher frequencies is limited.
Image 6.2.7 is given in the upper frequency limit for near-field measurements. It is the largest
Dimension of the source to use. It can be concluded that near-field measurements from approximately
300Hz to lose credibility.
Image 6.2.7: Upper frequency limit for near-field *)
see Annmerkung 2
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Image 6.2.8: Assessment of level correction for near-field measurements
Picture 6.2.8 allows a rapid assessment of whether in the near field, the microphonecould be controlled. If the speaker to be measured for example with 86 dB / W / m and an effective
specify tive membrane diameter of 8 cm, so we stayed in 1 cm distance of about 86 dB + 32
likely dB = 118 dB at one watt excitation level and thus are already in the maximum soundpressure measurement range of conventional electret microphones.
Note 1: In Figure 6.2.6 is the so-called "acoustic center" of the speaker
pointed out. This means that the selected reference level and the non Schallentstehungsort
inevitably coincide. This is evident in the analysis of impulse responses. With the
the ruler and measured the distance determined from the transit time of sound distance
often differ by a few centimeters (for determining the distance from the sound travel time see
Section 5.3.3, point 2). The resolution of this method is determined by the sampling rate of the
Sound card determines (48kHz = 7.2 mm, 96kHz = 3.58 mm)
Note 2: Using the source of the greatest dimension (space diagonal of the housing
ses) result of housing conditions not realizable measuring distances. As a compromise
can be either the biggest 3 times the diameter of the speaker, or for measurements in
Frequency response, at least 6 times the distance to the nearest edge of the housing to be taken.
Notes to the far field:
•
•
Measuring distance d> 3 * largest dimension of the source
The lower frequency limit f U depends on the maximum possible time slot (gate) of the room
(Cf. below)
Basically we have at far-field measurements, ensure that both the source and
Microphone can be placed as far as possible from reflective surfaces. In normal
Spaces is generally the ceiling height of the limiting dimension of approximately 2.50 m (land or
Ceiling reflection).
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Path of the floor or ceiling reflection:
DBoden / ceiling = 2 * √ ((d / 2) 2 + h2) [m]
Difference between direct sound and reflected sound:
Delta = DBoden / ceiling - d [m]
Skew:
Delta T = / c [s] where c = 344 m / s
Lower frequency limit:
fU = 1 / T [Hz]
Image 6.2.8: Measurement setup
To see reflections in the pulse diagram easier, you should before his
Measuring chamber analyze (see Figure 6.2.8). This image in 6.2.9 a little example.
Image 6.2.9: Analysis of the measurement
space
By the bill in the upper part of the image is the main reflections can be in the
Identify impulse response quite well. This is not always so easy because depending on the nature
the room (share and distribution of highly reflective or absorbent surfaces), the
Reflections more or less pronounced.
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Now a second example, which refers to Figure 6.2.10. With a ceiling height of
H = 2.20 m, a measured distance D = 0.53 m and 1.37 m measurement height h1 of results a
Acoustical DBoden / blanket for the floor or ceiling of reflection:
DBoden = 2 * ((0.53 * 0.5) 2 + 1.372) ^ 0.5 = 2.79 m
So 2.26 m longer than the path of the direct sound (measuring distance). This corresponds to a term of:
T = 2.26 / 344 = 0.0065697 = 6.5697 ms
and a lower frequency limit of:
fU = 1 / 0.0065697 = 152.2 Hz
In this room, and when said measuring distance we can give our Far-field measurements only
from 152 Hz up trust.
Other measuring distances for measuring height = half the room height are calculated in the
following table.
d [m]
0,030
0,060
0,120
0,240
0,480
0,960
h [m]
D floor / ceiling [m]
Delta [m]
1,100
2,200
2,170
1,100
2,201
2,141
1,100
2,203
2,083
1,100
2,213
1,973
1,100
2,252
1,772
1,100
2,400
1,440
T [ms]
fu [Hz]
6,309
158.5
6,223
160.7
6,056
165.1
5,736
174.3
5,150
194.2
4,187
238.8
Now let's look at the next pictures, as the frequency response with increasing
Measuring distance changes in the lower frequency range.
Image 6.2.10: Transition near field Far-field (0, 3, 6, 12, 24, 48, measuring 96 cm spacing)
From 6 cm, but no later than at 12 cm working distance first space influences are visible.
According to the instructions given above is for a measured distance <0.11 * dimension of
Sound source to be less than 1 dB of error. The largest dimension of the sound source in the above
example shown (FRS 8 in 2.0 liters CB) is about 26 cm. Thereafter, the measurement error should be
in the
Near field to remain at a measuring distance of about 3 cm below 1 dB.
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How does it look in the upper frequency range? 6.2.11 image shows the "windowed" frequency
outputs at different measurement intervals.
Image 6.2.11: Transition far field
Near field
Later than the transition from 24 cm to 12 cm are variations in the parallel and the 6
dB increase per halving of distance to watch. So we come gradually in the near field
(See also [9]).
What happens if we further increase the measurement distance. A few measurements
in a gym (27 x 15 x 5.5 m) at approximately 2.80 m measuring height and different measurement intervals
were made between 1.35 m to 3.79 m. To assess the properties of the measuring room
the reverberation time was also determined here. Image 6.8a shows the results: the mean
Reverberation time is about 3 seconds.
This results in a reverberation radius of about 1.40 m, which means that up to this distance measurement
Influence the room should be relatively low. We'll see!
Image 6.2.12: reverberation time (blue) / reverberation radius (red), a gym (27 x 15 x 5.5
m)
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With gate (red)
Without gate: 1 / October 3 (blue), 1 / October 24
(gray)
Burst decay
1.35 m
1.80 m
2.65 m
3.70 m
Image 6.2.13: Measurement of a solo in a gym with 20 different microphone distances
As is clear from the above - In the following table the other boundary conditions are
Known example - estimated. From the measurement conditions, time window of 8.6 to yield
12.8 ms.
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In Figure 6.2.15 are in the left row, the measurements with 1/24 octave (gray) and with 1/3
Octave smoothed (blue) without window (gate) to see. The red line shows the windowed
Measurements - that is to say while ignoring the measurement space.
In the right row is nice to see that with increasing distance measuring the influence of
Space is larger. The transition from the free field to the Hall field is well understood (Hall radius
about 1.40 m). Unfortunately, no measurements were made at shorter intervals around the measured to
demonstrate.
6.2.1 Determination of the reverberation time of the room characteristics
As already noted, takes the space in which we conduct our measurements, significant
Influence on the result. He changed the direct sound through echo and reverberation (see section 6)
and complicated by the isolated metrological description of the speaker.
Among the listed in ISO 3382 room acoustic parameters, reverberation time RT60 is one of the
important parameters. Unless the possibility of modification is, would be for measuring rooms
to strive for a very short reverberation time, for listening rooms in the home reverberation times are
recommended by about 0.4 seconds [5].
ARTA supports the determination of the reverberation time based on the requirements of the above
Standard. In carrying out the measurement of the ISO 3382 to observe the following
Boundary conditions required:
•
The microphone should be at least 1m from any reflective
Space and not too close to the source (speaker)
are positioned. The minimum distance from the source
can be calculated as follows:
V= Volume [m3],V
[M] c= Speed of sound [m / s],dmin =2
cT
T = Estimated reverberation time [s]
•
The sound source is a possible spherical
Have radiation. A special purpose
suitable source can be seen in the picture.
•
The microphone is omnidirectional have (see also
5.3.1).
•
The Excitation level should be 45 dB above the noise floor. Under normal living room
conditions a stimulus level> 90dB is therefore required.
•
To excite the room sufficiently, has the Excitation signal possible
be energetic. It is recommended to work with a sine sweep. In order to improve
signal to noise ratio additionally set in the menu "pulses response measurement " below
"Number of averages " 4 Center lungs.
Furthermore, it is important that the duration of excitation of the space much longer than the
estimated
Should be reverberation time.
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An estimation of the reverberation time can be performed by the following equation:
RT60 = 0.163 * V / A
V = volume in m3, A = equivalent sound absorption area in m2
A = Σ ai * Si
ai = sound absorption coefficient of the partial surfaces, Si = partial area in m2
Material
Unit
Carpet
Parquet
Wallpaper, plasterboard
Plaster, concrete, natural stone
Door, lacquered wood
Windows, insulating glass
Curtain
Shelf
Upholstered chair
Armchair
Sofa, love seat
m2
m2
m2
m2
m2
m2
m2
m2
Item
Item
Item
63Hz
125Hz
250Hz
500Hz
1000Hz
2000Hz
4000Hz
8000Hz
0,016
0,020
0,020
0,020
0,150
0,150
0,240
0,410
0,220
0,310
0,620
0,026
0,030
0,020
0,020
0,100
0,200
0,410
0,450
0,380
0,440
0,880
0,044
0,040
0,030
0,020
0,080
0,150
0,620
0,480
0,470
0,570
1,140
0,090
0,040
0,040
0,030
0,060
0,100
0,770
0,480
0,490
0,620
1,240
0,222
0,050
0,050
0,040
0,050
0,050
0,820
0,480
0,520
0,700
1,400
0,375
0,050
0,060
0,060
0,050
0,030
0,820
0,510
0,530
0,710
1,420
0,542
0,050
0,080
0,070
0,050
0,020
0,860
0,530
0,560
0,740
1,480
0,680
0,050
0,080
0,080
0,050
0,020
0,950
0,620
0,640
0,780
1,560
The above table shows some absorption coefficients for common "noise eater" in
relevant frequency band. For the estimation of the required excitation time
the calculation sufficiently at 125 Hz.
EXAMPLE:
Has a room with dimensions 4.9 x 3.8 x 2.2 m and a volume of 40.96 m3
Area shares with the following materials: 18.6 m2 carpet, 58 m2 of concrete / stone, 10 m2 shelf,
1.0 m2 window, door 3.6 m2, 2 upholstered chairs. This calculated
A = 18.6 * 0.026 + 58 * 0.02 + 10 * 0.45 + 1 * 0.20 + 3.6 * 0.10 + 2 * 0.38 = 7.46 m2
and
RT60 = 0.163 * 40.96 / 7.46 = 0.89 seconds at 125 Hz
The required duration of excitation should therefore be significantly longer than 0.89
seconds.
Figure 6.9 shows how in ARTA with the parameters outlined in red the duration of excitation set
can be. Where
Duration of excitation ≈ Sequence Lenght / sampling rate
With the available sequence lengths of 16k, 32k, 64k and 128k arise at 48
kHz sample stimulus durations of 0.33 s, 0.66 s, 1.33 s and 2.66 s That should be for normal
Living spaces to be sufficient. Who - for whatever reason - a longer excitation time
required can be achieved by reducing the sampling rate.
Note: For the determination of the absorption coefficients of materials by means of the in situ
ARTA measurement, see Application Note no. 8 [VIII].
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Image 6.2.1.1: Setting the excitation time
The impulse response of the room is shown in Figure 6.2.1.2. For orientation of the section is to
in the first room reflection, which is the area we normally for
Consider speaker measurements.
Image 6.2.1.2: Impulse response of the room
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By pressing the button the following menu is opened. The area in red are
all essential controls that we need for further evaluation.
Filtering
dB range
T60
Noise Tail
Login
Zoom
Scroll
Choice of the evaluated octave band or the entire frequency band (Wide)
Setting the Y-axis
Starts the calculation of the acoustic parameters. The result is below the
Displayed graphic
Consists of two adjustment variables:
-With the first variable determines which portion of the curve for evaluation
is used
-With the setting parameter is the second method of noise reduction
defined:
oTrunc - Thinks that the selected portion in the calculation does not
is taken into account
oSub - Thinks that the average noise level of the "tail" of
the curve is drawn
Edition of the report with the calculated room acoustic parameters
Horizontal zoom factor, Max or All
Move the graphic to the right or left
Image 6.2.1.3: Description of Controls
The analysis proceeds as follows:
1) Select the frequency band with "filtering"
2) Determine the evaluated part of the curve with "Noise tail". Here is a little
Try hip. The aim is the choice of the curve% by number and the method as
well adapted to the falling branch. The quality of this adjustment is referred to as
Correlation coefficient rto the next step directly below the graph
displayed. A correlation coefficient of r = 1 is optimal.
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Image 6.2.1.4: Evaluation by highlighting and marker
Image 6.2.1.5: Output of results
3) Determine the area by moving the cursor (yellow) and the marker (red)
to be evaluated. The evaluation is done by pressing the button T60.
4) Repeat steps 1-3 for all frequency bands.
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5) Output the calculated room acoustic parameters by pressing the Log
Results can be displayed as screenshots or as a CSV file. The CSV file
can be read directly into Excel, what the statistical analysis a little
relieved. Please make sure that the set-point in Setup under "CSV format"
will (see below).
Image 6.2.1.6 shows the statistical evaluation of three measurement positions with Excel. The red bars
show the simple standard deviation (spread) of the measurements.
Image 6.2.1.6: Statistical analysis of the individual results
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6.2.2 The automatic evaluation of the reverberation time
As of version 1.5 ARTA provides an automated evaluation of room acoustic parameters
gem. ISO 3382nd In "Acoustical Energy Decay" menu under "Automatic ISO 3382
Evaluation " 5 options available:
•
•
•
•
1/1 octave graphic evaluation
1/1 octave tabular analysis
1/3 octave graphic evaluation
1/3 octave tabular analysis
as well as the setup menu.
For the evaluation, only the
To activate desired menu item. For
the case of 1/1 octave graphic should
Results appear as follows (Figure 6.2.2.1).
Image 6.2.2.1: Graphical analysis for octave bands
To manipulate the graphical known options are available.
Furthermore, the results can be used as Overlay be stored.
In the "Parameters", all displayed room acoustic parameters as
Graphics are accessed (see picture left).
"Set" is possible the axes of the graph according to its own
Want to scale. Image 6.2.2.2 shows the possibilities offered. With
the "Update" button, a preview can be initiated.
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Image 6.2.2.2: Setup Menu for the graphics of the acoustic parameters
With the checkbox "Stepped Graph " , the nature of the graphical representation can be manipulated.
If it is enabled, the graphics shown as a band (bar) (see Figure 6.2.2.3).
Image 6.2.2.3: Graphical representation in third octave bands: line (left), bands (right)
As with the manual version, the results can be output as a table. It is
Note that is not included in the automatic evaluation of T60 (Figure 6.2.2.4).
Image 6.2.2.4: Tabular view
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6.3
Setup for acoustic measurements on loudspeakers
To the development of a speaker, in addition to knowledge and material, ARTA and a pair of ears
sufficiently. In order to reduce the development effort is now additionally
Simulation software (eg BoxSim, CALSOD) used. The virtual development process
reduces both the use of materials and the development time. Simulation results
are quite close to reality as a rule, but this require the consideration of some
When determining the frequency characteristics and impedance transitions with which the programs
to be fed. For this purpose the following are some hints that as not a recipe, but
Suggestion is to try to understand.
Simulation programs
From the large number of available simulation programs we look at the following
example, two representatives regarding the requirements of the measurement setup and the measured
data, with
where they are fed:
•
•
BoxSim
CALSOD
BoxSim offers the possibility of the individual speakers freely on the baffle to position (X,
Y-axis) and the entry site of the sound origin (Z-axis). The microphone is on
BoxSim virtually positioned at an infinite distance, hence there can be no angular error
passed between the individual speakers.
Due to this situation, it is necessary that each speaker on the road - or in
sufficiently large decency - to measure and test data as FRD or ZMA files into BoxSim
to import.
CALSOD is more flexible in this regard. It allows both the free position (X, Y,
Z) of the single speaker on the baffle and the microphone (X, Y, Z). This can
in principle, be mapped and measured and simulated each listening situation. The imported
Measurement data must also correspond to the conditions selected, or vice versa.
The following information is now to some of the variables considered in the measurement or
should be controlled.
Measurement
environment
Speakers are to meet the requirements of the listener in the selected listening environment.
Therefore, it would be logical for the loudspeaker development under these conditions to
measure. In the case of normal listening distances (1.5 to 4.0 m) and space dimensions (12 - 40 m2)
should
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However, you also have to realize that the so-obtained acoustic measurement results, the
Sum of speakers and room are.
If we look first to the development of a speaker independent from space - that is, in
Free field - limit, as we know from Section 6.1 that the measuring stick in space at some
specifies the way. Ceiling heights of 2.50 m, the limit of our available anechoic
Time window and thus determine the lower frequency limit (see Figure 6.3.1) and
Frequency resolution. Usual Hall radii of less than one meter indicate that when selecting
larger measurement intervals are not expected to more free-field conditions and therefore the
Room influence is dominant.
Image 6.3.1: Window length and lower cut-off frequency as a function of
Measuring distance for a room height H of 2.40 m (h = H / 2)
Test setup - angle error
These previously known limitations are further obtained by the measuring arrangement
incorporated. Picture 6.3.2 shows the geometry of a normal listening / measurement situation for a
Two-way speaker.
Angle to the horizontal
α = arctan ((MIK h - h HT) / D)
β = arctan ((MIK h - h TT) / D)
Image 6.3.2: Geometry of a normal listening / measurement
situation
For a real measurement, the microphone would, however, not without distress outside both
Position the speaker axes. Why, we are in the course of the following statements
. seen
In Image 6.3.3a are two different distance measurement positions for a two-way speaker
shown. The microphone is located on the axis of the tweeter, the woofer is each of
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A and the position B is measured. As reference, the measurement of the woofer on each axis
shows (A ', B'). The result is not necessarily a surprise: the shorter the distance measurement,
the greater the angle of measurement for the woofer and consequently the deviation from
Frequency response on axis. Let's go with this frequency response in a simulation program and
simulated for different distances, will inevitably errors.
Image 6.3.3a: Woofer on each axis (A '= green, B' = red) and Axis HT (A = black, B = blue)
Image 6.3.3b shows the measurements for the tweeter. It is clear that, for measuring distance 60cm
occur as early as 1.5 kHz angle error.
Image 6.3.3b: Tweeter on each axis (A '= green, B' = red) and Axis TT (A = black, B = blue)
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However at 150 cm distance, the angle measurement error - for the conditions shown here be tolerated, since the deviations from the reference axis to start only at about 10 kHz,
ie 1.5 to 2 octaves above normal transition frequencies.
Order to estimate the angular error for other measurement parameters is in Figure 6.3.4
Connection between chassis and measuring distance and the respective measurement angle shown.
Image 6.3.4: Measuring distance as a function of angle and distance Chassis
Example: How big is the measuring distance D must be at least when at a distance d of 21 cm
α between the two speaker chassis measuring angle should not be greater than 10 °?
If we look at the intersection of 0.21 m spacing chassis with the 10 ° line, results a
Minimum measuring distance of about 1.18 m.
We see this measuring arrangement required to avoid a large angular errors
Measuring distance, which might then quickly to the right to compliance with the free-field conditions
contrary.
Geometric delay differences
We now devote our attention to another point, which also consists of the
Measuring arrangement results. 6.3.5 shows that in addition to different measuring angles also
different measuring distances and times are taken into account for the sound
Distance Microphone - LS
DHT = sqrt ((h MIK - HT h) 2 + D2)
DTT = sqrt ((h MIK - TT h) 2 + D2)
Path difference
DELTA.D = (DTT - DHT)
Skew
At = DELTA.D / 344 m / s
6.3.5: Phase shift due to different maturity
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The following table shows the red columns, the measurement conditions for the examples
Image from 6.3.3a and 6.3.3b image. At 60 cm a DELTA.D path difference of 1.85 cm, which
a duration of 0.054 ms corresponds difference At. When measuring distance of 150 cm is reduced
the path difference of 0.75 cm and a time difference of 0.022 ms.
The time difference is equivalent to a delay (delay), the frequency with which a
continuously increasing phase shift corresponds to:
dPhi [°] = delay [m] * [Hz] / speed of sound [m / s] * 360
In the case of a conventional crossover frequency of 3000 Hz corresponds to the delay of 1.847 cm
a phase shift of
dPhi [°] = 0.01847 [m] * 3000 [Hz] / 344 [m / s] * 360 = 57.98 °
relative to the tweeter. The simulation in Figure 6.3.6 gives an impression of the influence
these 1,847 cm under the stated conditions on idealized speaker with Linkwitz Riley
Filter 2 Order have.
Image 6.3.6: Effect of transit time differences (left without, right with delay)
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The Schallentstehungsort (SEO)
So far we have assumed that the Schallentstehungsort in speaker on the chassis
Is the level of the baffle. Unfortunately, this is not quite the reality.
After excitation by a signal, the speaker cone, driven deflected by the
Voice coil and air produces sound. This deflection is not at all frequencies for
all sections of the same membrane (piston-like radiation), but any deformation
and resonances within the membrane. These processes require some time before they are from the place
the force, the voice coil, have propagated to the individual membrane sections
and then there will be radiated as sound. The running time depends on both the dimensions of the
Membrane as well as the properties of the membrane materials. It is easy
imagine that this process will be frequency-and location-dependent. Furthermore, it can easily
show that unlike the model of the point source in a real speaker does not
all membrane sections are the same distance from the microphone.
In a real speaker so do not expect it to be like a point source
behaves. The so-called Schallentstehungsort (SEO) will not be determined as a fixed point,
but depends on the frequency wander. One of the common suggestions, the position of the voice coil
to accept as SEO, therefore should not be entirely accurate. Overall, it is located
extremely complex relationships that repeatedly investigated in several publications
were. Also in the manuals of simulation or speaker measurement software will be
find it. The advice given to determine the so-called sound development site (SEO)
thereby cover the spectrum of rough approximations from up to scientific treatises.
Image 6.3.7: phase shift due to different runtime with consideration of SEO
The possibilities mentioned in the literature and methods of determining the SEO to the
are not deprived of course interested reader [17] - [21], but we should not
forget that
1 the SEO is only one aspect among others, for in the preparation of the data
Simulation programs Note
2 do not depend on the absolute values of the simulation of SEO, but on the
relative differences between the chassis used.
3 The crossover also has a significant influence on the time behavior.
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Methods for the determination of SEO
A simple nmap to determine the SEO is shown in Figure 6.3.8. The microphone is
on the axis of the measured loudspeaker (woofer here) positioned at a distance d. The distance
from the microphone to the level of the sound wall (reference level) is possible with a tape measure
to measure accurately. In our example, there are exactly 60.0 cm.
Woofer
Tweeter
Image 6.3.8: Determination of SEO (distance method - pulse maximum)
Now we measure the impulse response of ARTA in dual channel mode and determine therefrom the
Duration or the distance to the pulse maximum. If the menu item View, select gate
Time is selected, the distance can be read off directly below the graph (see Figure 6.3.8,
right).
Woofer
At = 1.833 ms DIMP = 1.833 * 34.4 = 63.07 cm
SEO Δ = d - DIMP = 60.0 to 63.07 = -3.07 cm The SEO of the woofer is 3.07 cm behind the
Baffle plane.
Tweeter
At = 1.771 ms DIMP = 1.771 * 34.4 = 60.92 cm
SEO Δ = d - DIMP = 60.0 to 60.92 = -1.47 cm The SEO of the tweeter is 0.92 cm behind the
Baffle plane.
Both SEO thus differ by 2.15 cm.
Note 1: As of version 1.4 which was ARTAum
Extended overlay function for impulse responses.
In this way two impulse responses compared
be and the time differences directly by
Cursors and markers can be determined.
In this context, the
Possibility of manipulation of the Markers by means of the
right shift key and to mention arrow keys
because it allows a better positioning of the
Marker.
Note2: Version 1.4 offers ARTA in FR2
Fashion automatic distance calculation
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Joseph D'Appolito [2] proposes a method to determine the difference between two SEO
requires two measurements and a little math. The measurement geometry is shown in Figure 6.3.9
shown. Both are measured from a loudspeaker and microphone position of the
Impulse responses - similar to the example above - the distances d1 and d3 determined. Then calculated
to d2 (which is the distance that would result if the woofer at the point of
Would have been tweeter) as follows:
_______
d32 = √ d2 - h2, and Dd = d2 - d1
Tweeter d1
Woofer d3
Image 6.3.9: Determination of SEO (method d Appolito))
With d1 = d3 = 60.92 cm and 64.50 cm arises at a distance h = 15.0 cm d2 = 62.73 and thus
differ the SEO, this method is Dd = 1.81 cm.
The resolution of these methods is as already mentioned in Chapter 6.2 of the sampling rate
the sound card is determined (= 7.2 mm 48kHz, 96kHz = 3.58 mm).
A third method is represented by the group delay function. For this purpose, in the
Impulse responses of the windowed room reflections and then displayed as Group Delay
(See Figure 6.3.10). The cursor is set to the selected transition frequency and the
Delay read below the chart.
At = 1.813 ms
d = 1.813 * 34.4 = 62.37 cm
At = 1.765 ms
d = 1.765 * 34.4 = 60.72 cm
Image 6.3.10: Determination of SEO (Method Group Delay)
This method results in a difference of 1.65 cm of SEO. If the two Group Delay
Gradients by means of the overlay are displayed in a function chart, the method slightly
illustrative (Figure 6.3.11)
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Image 6.3.11: Determination of SEO (Method Group Delay)
Other methods for the determination of SEO - for example using an approximation to the minimum-phase
Course or integration of simulation programs - are from the literature [17] - [21]
found. Who wants to try everything, should take some time.
For those who have worked his way up to this point, the disillusionment. Even the
careful investigation and consideration of all the variables described above does not lead directly to
Soft perfect, because the crossover itself influences the time behavior.
So bring a low-pass signal delay. This delay is with increasing
Filter order and decreasing frequency and is always greater separation at low frequencies - in
Compared to the geometric displacement of SEO - be dominant. Course development and
therefore remains a "work of art"
Image 6.3.11: Signal delay by 18dB lowpass
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6.4
Scaling and joining of near-and far-field measurements
For further processing in simulation programs is a complete frequency response
(Phase and amplitude) is required. To be a near-field and far-field measurement
together (see also section measurement in a reflective environment).
For the generation of the overall frequency response are some steps to go through, the basis of the
Two examples are shown. For a
•2 liter closed box with a full-range loudspeaker Visaton FRS8
•and an 8 liter bass reflex speaker with a 5 "chassis
be reworked with ARTA measured frequency responses.
6.4.1 Closed Box
1) Measure and store the Nahfeldfrequenzganges
Image 6.4.1: Impulse response in the near
field
Put the cursor (yellow line) to the beginning of the first pulse to a correct
To obtain phase relation. Attention, if the cursor is placed too close to the pulse peak,
Information can also be lost. It is better to keep some distance and
then the difference to be corrected by a delay. So the cursor (left mouse button), about 1
ms place before the first pulse, the marker (right mouse button) on the right
Set and with pulse maximum, Get 'take over the delay in the top menu bar.
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Evaluate the impulse response by
from.
Set in, Smoothed Frequency resonse ' via
Menu, view 'the view Magn + phase' a.
The next picture is the frequency and phase response of the
Speaker to see in the near field.
In a membrane diameter of approximately 6.4 cm of Nahfeldfrequenzgang is valid up to 900 Hz
(See Figure 6.4.2). To make this clear, the cursor was set at 900 Hz.
Image 6.4.2: Frequency response in the near field. "Scope" marked by the cursor.
2) Correction of Nahfeldfrequenzganges on the measurement distance of the far field. Here offers ARTA
two options:
A) In, Smoothed Frequency Response '
on the 'Edit' menu 'Scale Level'
The correction value for the level (2 Pi)
calculated with
Correction (FF) = 20 * log (a / 2d)
a = radius of the membrane, d = measured
distance
with a = 3.18 cm and d = 48 cm results
Correction (FF) = -29.6 dB
B) In the above ARTA Main Menu 'Edit'
, Scale 'in the time domain
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3) Implementation of the baffle step correction
A special feature of ARTA is the correction of the so-called "Baffle Steps" (for more details
this can note in the ARTA Application No. 4
be found [IV]).
For this purpose, under 'Edit LF diffraction box' the left
dialog box shown open:
Here are the shape (square, rectangular, spherical)
and the dimensions of the input box.
After pressing OK Figure 7.2b should be visible.
This curve is stored as an overlay.
Image 6.4.3: Frequency response in the near field with baffle step correction
(black)
4) Download or measuring the far-field frequency response
Now we open the file with the impulse response of the far-field frequency response and set the "gate"
(Yellow line = left mouse button, red line = right mouse button). Very nice to see the densely
lying together reflections from floor and ceiling (speaker stands in approximately half
Room height).
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Image 6.4.4: far-field impulse response (48 cm) with gate
Below the graph, the length of the gate in ms is shown. Short Crosscheck: 5.146 ms
correspond to 1.77 m sonic time. This corresponds exactly with the theoretical considerations of the
Example match in the previous chapter.
After evaluation, you receive the following preliminary medium combination of local and
Far-field frequency response (Figure 6.4.5). It can be seen that the level adjustment works quite well
has.
Image 6.4.5: "Rohfrequenzgang" near and far field
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Now you can determine the frequency at which the transition, or rather want to cut.
In the example shown here, offers a frequency of about 240 Hz.
Now we move the cursor (yellow line) to the desired
Transition frequency, and go to the menu 'Edit'.
With the command, merge overlay below cursor 'is as the
Overlay defined Nahfeldfrequenzgang left of the cursor to the
Added far-field frequency response and the far-field frequency response
left of the cursor deleted (see Figure 6.4.6).
If you are in the menu overlay 'any remaining overlays
delete, you will see the remaining overall frequency response.
Overall, the transition looks pretty clean, this also applies to
phase.
Image 6.4.6: Overall frequency response (quasi-free
field)
5) Export the sum frequency response
Using the menu File , Export ASCII ' you can change the frequency response of the assembled
Export further processing in simulation programs.
There are two options:
•Export as ASCII file with comments for measuring
•Export as FRD format (ASCII without header and comments)
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Image 6.4.7: Export of the overall frequency response
If you want the ASCII export, select the query shown above Cancel '
for the export FRD select 'OK'.
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6.4.2 Bassreflexbox
For completeness, we will look at a small bass reflex speaker in the near field. Here
we have to consider two sources of sound, the speaker cone and the reflex tunnel.
Up to this expansion, the joining of the near-and far-field frequency response runs
similar to the example shown above.
In the chosen example, the diameter of the
Reflex tunnel (DP) and the effective diameter of the
Speaker diaphragm (DD):
DP = 4.80 cm
DD = 10.20 cm
Picture 6.4.8 shows the positioning of the microphone for the
Membrane and the reflex tunnel. If measurement error <1 dB
should be, the measuring distance should not be greater than (see
[03] or Section 6.2):
Reflex tunnel
Membrane
0.26 cm
0.56 cm
Image 6.4.8: Positioning the
Measurement microphone
Picture 6.4.9 shows the impulse responses of the membrane (black) and the reflex tunnel (red). The
Pulse of reflex tunnel comes with approx 0.72 pm (24.72 cm) delay at the microphone.
Image 6.4.9: impulse response of membrane (black) and reflex tunnel (red)
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Image 6.4.10: Membrane and reflex tunnel in the near field level without
correction
Image 6.4.10 shows the impulse response of the membrane and the bass reflex tunnel in the near field.
According to
6.2.7, the near field image in the used 5 "chassis (RD = 5.1 cm) to about 500
Hz can be used. To hide the higher frequencies was omitted.
The positioning of the microphone was shown in Figure 6.4.8. Since reflex tube and membrane
have different radiating surfaces, we need to make a correction level.
Calculation of correction factor
PNF = PD + (SP / SD) ^ 0.5 * PP
Level PNF = Near field PD = Level of membrane
PP = Regular reflex port (port =)
Port areas / membrane:
Sp = 18.01 cm ^ 2
SD = 82.00 cm ^ 2
SD = diaphragm, SP = reflex tunnel
Image 6.4.11: Enter the scaling values
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Image 6.4.12: Membrane and reflex tunnel in the near field with level correction
Image 6.4.12 shows the level corrected frequency response of the reflex tube, together with the
Frequency response of the membrane. It is very nice to see that the reflex tube and outside the
desired work area radiates sound.
Image 6.4.13: Total frequency response (black) and of membrane reflex tube
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6.4.13 shows the image with "Load and Sum " determined overall frequency response in the near field. It
can
be up to about 500 Hz is used.
a) far-field frequency response calibration,
b) Near and far field without correction level
c) near-and far-field, with level correction
d) Close with "baffle step compensation"
e) near-and far-field, merge overlay
f) quasi-free-field frequency response
Image 6.4.14 development of the quasi-free-field frequency response for a bass reflex
To complete the "quasi-free-field frequency response" is missing now still adjusting to the
Far-field measurement. Image Image 6.4.14f 6.4.14a to show the whole process in steps. The
Level adjustment - shown in graph c -, according to the described in section 6.4.1
Method performed with a final "optical" fine tuning. It should be remembered that the
Nahfeldfrequenzgang using the "LF Box Diffraction Function "must be corrected (panel
d). Then, near-and far-field frequency response with the "Merge Overlay " Function - in
this example, at about 240 Hz - together (panel e and f).
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Level matching in the same volume flow
In this method, I was made aware of Fabian Reimann. Thank you Fabian!
Image 6.4.15: LspCAD simulation
The method assumes that at frequencies well below the tuning
Volume flows and thus the levels are almost the same (see Figure 6.4.15). It is not
be mentioned that - especially in low tuned boxes - can be difficult,
still below the tuning frequency to measure clean sequences (see also Chapter 6.0.2, 6.0.3).
Image 6.4.16: Volume flow method implementation in ARTA
Consequently, the level of reflex tunnel (Figure 6.4.16, the left blue arrow) is to reduce the extent
until in the lowest frequency range of the level of
Membrane coincide (see Figure 6.4.16, right
Part of figure).
In the chosen example, the required
Reduction around -6.5 dB. Thus, the level of
Reflex tunnel to 10 ^ (-6.5/20) means "Pir Scaling "
to correct. The rest of the procedure followed by the
standard procedure described above.
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6.5
Load and Sum
The "Load Function and Sum " something is briefly described in the current manual and additional
hard to find. That one in ARTA with overlay any number of individual frequency responses
can cache has already been described (see Figure 6.5.1).
Image 6.5.1: Preparation of 1 to n single-frequency responses with ARTA
But what if we measured or imported from a single frequency responses
Want to form sum frequency response?
Image 6.5.2: The ARTA File menu
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There are two options:
•
•
Export as ASCII data and make the summation in a simulation program
"Load and sum" directly in ARTA
"Load and Sum " -To find the File menu - loads a previously saved file and adds PIR
it to the current signal in the store. One can with ARTA signals in the time domain
add up. The first does not sound so exciting, but it is still a useful
Function, for example a life may facilitate the development of switches.
In detail, it works exactly as described in the original manual:
•
•
•
Measure or loading the first PIR files (for example HT)
a previously saved file with PIR "Load and Sum" load (eg TT)
Total pulse mitauswerten.
The result should be the sum of the frequency response to be (see Figure 6.5.3).
Image 6.5.3: "Load and Sum" with two single frequency responses
Oops, what's that? That should actually look different! This is explained by the fact that ARTA
the newly loaded impulse always summed up the data in the memory. So be careful not
always be seen immediately the error.
How it should work for sure? Also helps here (see Figure 8.6), the File menu.
The "New" clear the memory and start the virgin is nothing in the way:
•
•
•
A file (for example DD) charge as normal with
"open"
Load File B (for example HT) with "Load and Sum"
Evaluate completed (see Figure 8.4)
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Image 6.5.4: summation with ARTA (memory previously deleted)
Oops the second. Apparently, no attention was paid to the correct polarity.
How then can for example the tweeter be reversed? Same procedure!
•The "New" clear the memory,
•File B (HT) and normal load with "INV" invert the phase.
•
•
Load file A (TMT) with "Load and Sum"
Evaluate completed (see Figure 6.5.5)
Image 6.5.5: Load and sum with inverted phase (HT)
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6.6
Working with targets
Targets - or even objective functions - are in many situations of everyday useful measurement, eg with
the development of crossovers, the Declaration of baffle effects, or
Confirmation of simulations with measurements.
Targets are in ARTA in the "IMP
Smoothed Frequency response " and in "FR1"
and "FR2" and there are provided in each case
Find menu "overlay" (see left).
It can using "Generate target response "
common standard filter functions are generated,
or "Load target response " any
Functional course load as a foreign file. It
the frd and txt formats are accepted. The
Function "Delete Target Response " deletes All
displayed targets.
Standard filter functions as a target
The menu for the mapping of standard filter functions "target Filter Response " is
"Generate Target Response " reached (see Figure 6.6.1). Means "Reference passband
sensitivity " is the level of the
Target feature set.
About the choice of "filter type "
(High-, low-, band-pass), the
Filter type (Butterworth, Bessel,
Linkwitz) including the Filterordtion and the Übergangsfrequenzen (crossover Frequencies)
the target function may be
true. By confirmation of
"Ok", the target function
plotted.
The process can often
be repeated (see Figure
6.6.2), all generated targets
remain until they actively
"Delete target response " deleted
are. A selective deletion
individual curves is not
possible.
Image 6.6.1: Menu "Target Filter Response"
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Image 6.6.2: Example targets: Filter functions of different order
Standard filter functions as a guide in the development of crossovers
be useful. It specifies the desired objective function and attempts by varying the
Filter components to this approach (see Figure 6.6.3).
Image 6.6.3: Target and measured frequency response of a switch branch
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In the measurement mode "FR1" and "FR2" this can also be done "online". In particular
Using variable inductors and capacitors is the method very effective.
Image 6.6.4: Target and measured phase response of a switch branch
The target function is not only on the frequency response, but also to the phase response
applicable (see Figure 6.6.4). In this context, nor can a more interesting way of
Target functions in conjunction with the "Delay for Phase Estimation " shown
be (found under Edit delay for phase estimation).
By inserting a delay, the measured phase can be approximated to the target function
(See Figure 6.6.5). The original data are not changed, the data export is the
Delay inserted is recognized.
Image 6.6.5: Target phase and measured phase response of 0.0ms, 1.0ms delay and 1.3639ms
Any target functions
If the desired objective function can not be mapped using the standard filter functions, you can
arbitrary functions using the "Load Target Response " be imported. As far as known, are
all exports from simulation programs with the extension txt, frd or zma accepted.
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In the first example, the common representation of frequency-and impedance response is shown
are. For it is in an existing frequency response via the menu "overlay" and
"Load Impedance overlay" LIMP loaded either a file or a txt or zma file.
Loading impedance overlays a second Y-axis is opened. This is by
the menu "Graph setup" to manipulation.
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Image 6.6.6: Joint representation of frequency-and impedance response
(Top) tweeters, (below) bass reflex cabinet
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In the second example, the simulation of a bass reflex speaker with CALSOD is by measuring
be verified. The simulation includes a special feature, the speaker and
Housing parameters are determined from the impedance measurement of the prototype. Withhold
Waldman
has the method presented in 1993 at the AES Convention in Munich and CALSOD
implements [28]. Picture 6.6.7 shows the impedance curve before and after the parameter optimization
with CALSOD (measurement · · · · · · · ─ ─ simulation).
Image 6.6.7: Determination of TSP from the impedance curve of a bass reflex speaker
with
CALSOD. Before (left) and after parameter optimization (right).
Image 6.6.8: Parameters determined from the calculated frequency
response
Picture 6.6.8 shows the calculated frequency response of the measured prototype. For the
required parameters were calculated from the impedance response. 6.6.9 and 6.6.10 Image Image
show the comparison between measured data (black) and simulation data (red) for two
different housing sizes and votes.
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Image 6.6.9: Comparative simulation (red) with measurement (black) for VB = 18 ltr
Image 6.6.10: Comparative simulation (red) with measurement (black) for VB = 31 ltr
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In the third example, the simulation of a 1 m long transmission line (TL) is determined by measuring
be verified. The simulation was performed using AJHorn 5.0 (www.aj-systems.de) by Armin Jost
made and then exported the data (see Figure 6.6.11).
Image 6.6.11: Simulation of a 1m long TL with AJH
The near-field measurements of membrane and at the end of the TL were calculated using the
Volume flow method combined (see chapter 6.4.2).
Image 6.6.12: Imported target function (red) vs. Measurement (yellow)
6.6.12 image shows measurement and simulation in a diagram with surprisingly close agreement
determination. In Figure 6.6.13 of the influence of a damping at the end of the TL is examined. Are in the
image
AJH in the set parameters shown.
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Image 6.6.13: Influence of damping at the end of the TL
By comparison of measurement and simulation one gets a sense of the impact of
initiated containment measures in relation to the AJH variables β1 and β2.
In the fourth example is the simulation of the influence of a baffle with EDGE
(Www.tolvan.com / edge) can be verified by measurement. EDGE also provides an export function
the simulated data to.
Image 6.6.14: Effect of baffle: EDGE simulation (red), measurement (gray)
6.6.15 In the picture a little gimmick is completed, the measured data (blue) were compared with the
data
EDGE simulation (green) corrected. The red curve then represents the sound pressure curve without
Baffle effects represent, which roughly corresponds to the measurement on a standard baffle. It applies
but only for the example realized measurement position!
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Image 6.6.15: Correction of the baffle influence on the measurement position
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6.7
Electrical measurements on crossovers with ARTA
For the development of crossovers are also electrical in addition to acoustic measurements
Measurements to test the effect of the respective circuit interesting. For this reason,
should be given here some hints and examples. The chapter does not explicitly
the claim of the treatment of the development of crossovers.
As already mentioned several times, generally when the electrical measurements Caution
attached. So before measuring with a multimeter the voltages at the crossover and
then through a voltage divider (see Chapter 5) to adjust the level for the sound card. Image
6.7.1 shows the experimental setup for electrical measurement. The "probe" with a protective function
is marked in red in the left image, the right picture you can see the practical implementation.
However you can also use the microphone input of the ARTA MessBox the probe.
Then comes - depending on the input impedance of the card - still about 0.5 dB attenuation of the effect
added to the voltage divider.
Image 6.7.1: Measurement setup for electrical measurements on crossovers
Regarding the dimensioning of the voltage divider is one of the values from Figure 2.6 (see
Section 1.4) under normal measurement conditions on the safe side. In one watt
Input power is the power at 8 ohms U = √ 1.8 = 2.83 V. Since the 1:10
Voltage divider is almost too much of a good thing.
Who are the voltages and currents in standard crossover a little more in detail
want to see, which is the program "PassFil" from the homepage of Bullock & White
(Http://users.hal-pc.org/ bwhitejr ~ /) is recommended. The following example shows the means PassFil
calculated voltage curve on the components of the high pass of a two-way crossover at 15
Watts of power. We see, at 15 watts, it is with our 1:10 voltage divider getting tight.
Image 6.7.2: Voltage curve at the crossover components marked with PassFil
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In the following text (6dB high pass) to be shown by an example, such as electric
Measurements can support the development of crossovers.
Picture 6.7.3 shows the frequency response of the tweeter without (red) and with (blue) crossover. The
crossover
is very simple, it consists only of a 6.8 uF capacitor. Both during the
Amplitude response and the measured slope not really look for a
First-order filter. The high acoustic slope is due to the superposition
the electrical high-pass filter with the acoustic of the tweeter. Actually, it should only
To be 12 dB + 6 dB = -18 dB per octave due to the acoustic parameters of the filter ((Q = 1.6,
f = 1400Hz) it is a step in the direction of 24 dB / octave (see simulation in Figure 6.7.4). This
one may evaluate the obviousness of the formulaic treatment of crossovers.
Image 6.7.3: Frequency response with / without 6dB crossover
Image 6.7.4: Simulation 6dB crossover with resistive termination
(___ HT (Q = 1.6), ___ 6dB filter, filter .... + HT)
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If the difference is made up of two acoustic frequency responses (see Figure 6.7.5), we obtain
themselves - except for the discontinuity at about 1.2 kHz - but apparently a filtering effect of 6
dB / octave.
Image 6.7.5: Difference with / without 6dB Crossover (acoustic)
We now take the measurement signal from the microphone is not, but - as shown in Figure 6.7.1 - about
the sensor directly from the switch (see Figure 6.7.6), the image is clearly the
electrical filter effect is 6dB. The discontinuity at 1.2 kHz seems to come from the
Interaction of the impedance of the tweeter (gray) with the condenser of the crossover.
Image 6.7.6: Amplitude variation with 6dB Crossover (electric)
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What happens if we have the resonance of the tweeter with an RLC
Member smooth? The adjacent circuit showing the structure of the extended
Crossover.
Picture 6.7.7 shows the acoustic result compared to the control without
RLC element. The interaction of the resonance of the tweeter with the
Capacitor seems to be almost eliminated.
The acoustic difference between the amplitude curves with / without Soft is now approaching
clearly the course of a first-order increases (Fig. 6.7.8) filter.
Image 6.7.7: Amplitude variation with / without RLC element (acoustic)
Image 6.7.8: Difference with / without crossover 6dB + RLC element (acoustic)
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We make again the opposing electric control. 6.7.9 image shows that the peak at 1.2 kHz
was significantly reduced, but not eliminated. It appears that the sizing of RLC
Correction term is not yet optimal, what is the impedance curve (gray) is also shown.
Image 6.7.9: Amplitude curve 6dB + RLC element (electric)
What can now be achieved by further optimization of the RLC corrector? To
we look at picture 6.7.10. The electrical profile of the filter curve (black) corresponds now
almost the 6dB target. The elevation of the amplitude curve (red) in the
Resonance frequency is disappeared.
Image 6.7.10: Amplitude curve 6dB + optimized RLC member (electric)
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Image 6.7.11 (acoustic) and image 6.7.12 (electric) show the summary of the optimization
the RLC member. The values of the RLC element were as follows:
ZuordnungR in ohms L in mH C in uF
Image 6.7.11 / 6.7.12
Blau8, 21,1727,0
Grün8, 21,1717,0
Rot8, 21,1713,3
Image 6.7.11: Amplitude and impedance curve RLC element optimization (acoustic)
Image 6.7.12: Amplitude and impedance curve RLC element optimization (electric)
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This little foray into the electrical measurement on crossovers - fixed to a very
simple example - shows that such measurements to support additional clarity
can bring, since the hidden influence of the measuring space and other uncertainties
are. So it is worthwhile to have the "probe" image from 6.7.1 in the measurement case.
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7
Special measurements and examples
7.1
Measurement of harmonic distortion sine
By means of a by Farina [11] proposed method for the rapid simultaneous sine can
Determination of frequency response and harmonic distortion are used. However, it is not
fully tested method because it provides results in which not all other types of distortion,
Reflections induced by noise or artifacts are isolated from the harmonic components.
However, the method is useful because it enables faster inspection of the
Structure and frequency characteristics of harmonic distortion than with other methods
is possible. To achieve good results, the measurements should be in rooms with little
Be performed little reverb and impulsive noise [13].
Hereinafter, the frequency response and the determination of harmonic distortion by means of the
described automated method that is implemented on the ARTA version 1.3. For
Measurement, the user has to perform the following steps:
1 In the sweep mode, the single-channel enable (Dual channel measurement mode
disable). Please note that for the calibrated measurement procedure according to Chapter 3.3
must be run.
2 Set the checkbox 'Center peak of impulse response ' (Figure 7.1.1)
Image 7.1.1: ARTA setup menu for the measurement of the impulse response with sine
3 Performing the measurement (Record). The length of the excitation sequence must be at least 64k
his or greater. The measured impulse response should look something like in Figure 7.1.2
shown.
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Image 7.1.2: Enlarged impulse response (IR). Show the sections marked in red
the gates of the linear distortion and the IR-induced IR for the second, third and
fourth harmonic.
4 The cursor several samples before the peak of the impulse response set (less than 250 samples)
and
5 the key combination Shift + F12 . Press
ARTA then automatically processes the evaluation procedure and displays the results in
the new analysis window "Frequency Response and Distortions "(Fig. 7.1.3).
Image 7.1.3: 'Frequency Response and Distortions' window
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The upper curve shows the frequency response, and the lower curves show the harmonic
Distortion 2, 3 and 4 order.
The manipulation of the graph is similar to that of the ARTA "Smoothed Frequency
Response "window. The complete setup menu is obtained by the command 'View
> Setup ' or by right-clicking on the graph. This opens the dialog box
'Magnitude / Distortion Graph setup ' as shown in Figure 7.1.4.
Image 7.1.4: Dialog box for the graphic setup
Picture 7.1.5 shows a comparison between the method and Farina STEPS in single channel
Fashion for 4 different levels. Otherwise, all boundary conditions are identical. Both
Klirrverläufe and Klirrpegel hardly differ, but should the introductory
Note be considered.
It is striking that in the right sub-images (STEPS) with decreasing excitation level and the
measured acoustic level decreases. The explanation is that ARTA with reference levels
works while STEPS identifies the absolute level in single channel mode.
The single-channel measurement is therefore well suited for the determination of the absolute level at the
microphone
and thus to establish a possible SPL exceeded or "excessive demands"
the microphone.
More to the distortion measurement, see Section 9.2 "Amplitude and distortion measurements with
STEPS ".
Note: From version 1.4, the measured data for further processing in other can
Programs are exported. As usual, there is a ASCII and CSV export.
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Farina
STEPS single channel
1dB
- 3 dB
6dB
- 12 dB
Image 7.1.5: Comparison vs. Farina. STEPS at 4 different levels
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7.2
Sound level measurements with ARTA
Music is often not perceived as beautiful because they always associated with noise. This quote from
Wilhelm Busch comes to the chagrin of listening - or their neighbors - in rented apartments
too often to bear. What in what circumstances is now loud or quiet, in guidelines,
Technical guidance and standards set (eg Directive 2003/10/EC or DIN 15905-5:
Entertainment technology - sound - Part 5: Measures to prevent the risk of hearing impairment
the audience by high noise levels electroacoustic sound systems).
The measurement of sound levels as well as the equipment required for this purpose is defined in IEC
61672-1:2002 defined. As of version 1.4, a virtual sound level meter is integrated in ARTA,
on Neudeutsch also Sound Pressure Level SPL meter or meter called. The structure or the
How the ARTA sound level meter is shown in Figure 7.2.1
Image 7.2.1: Block diagram of the integrating sound level meter
The microphone signal is input to the amplifier to
Overload indicator, indicates the status of the input amplifier and the A / D converter of the
Sound card displays.
From there, the signal goes to the weighting filter A, C or Z (see IEC 61627-1 or image
7.2.2) wherein Z "unweighted" or "linear" is. This review filter for RMS
Level measurements used in the case of peak-level measurements, only the C-weighting filter
used.
In the next stage, the signal is squared and then goes into the integrator or the peak
Detector. The signal is then square-root and logarithm, and finally the display
appears as noise in dB.
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Image 7.2.2: Weighting filters A, C, Z
The sound level meter in ARTA is with the command "Tools
A window will appear as shown in Figure 7.3.3.
SPL Meter " activated. So that
Image 7.2.3: SPL meter window in ARTA
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The controls in the SPL Meter window have the following meaning / function:
Leq - Linear averaging:
LAeq current value of Leq in uppercase.
Time Time relative to the beginning of the measurement.
Weighting Choice of the weighting filter A, C or Z (lin).
Hours, Minutes and Seconds Definition of the duration of the measurement
(There are a maximum of 24 hours, 59 minutes and 59 seconds allowed).
SPL - Exponential averaging:
LAS current value of the time-weighted SPL (with weighting filter A).
LAmax maximum value of the time-weighted SPL for the entire measurement period.
Lamin minimum value of the time-weighted SPL for the entire measurement period.
Weighting Choice of the weighting filter A, C or Z (lin).
Time integration Choice of the time weighting F (fast), S (Slow) or I (pulses).
Peak Level
LCpk current peak level (C-weighted, time interval 1 s).
LCpk, max maximum peak level (C-weighted, for the entire measurement time).
Audio Devices
Sampling rate Choice of the sampling frequency (44100, 48000 or 96000 Hz).
Rec / reset the measurement starts or resets all values to zero (reset).
Stop the measurement stops.
OK closes the "SPL Meter" window.
Peak meters dBFS shows the current peak operating levels relative to full scale value of the
Sound card in dBFS.
Record SPL history enabled the logging data in graphic mode (level recorder). There are
5 values recorded Leq, LSlow, lfast, Lpeak and Limpulse.
The manipulation of graphic works, mutatis mutandis, as in other areas of work of ARTA.
The fine adjustment of the graphics in the menu 'SPL graph setup ' as shown in Figure 7.3.4.
Image 7.2.4: SPL Graph Setup
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The controls have the following meaning / function:
Magnitude axis
Top Magn (db) sets the maximum value of the Y-axis
Magn range (dB) sets the range of the Y-axis
Time axis
Graph max - Definition of the upper time limit
Graph min - Definition of the lower time limit
All information in relative time values (no time to enter).
Curves show
Leq, LSlow, lfast, LPeak, LImpulse enables / disables writing to corners.
Thick lines plot - Enabled line style, thickness.
Show local time - On the time axis on time.
Graph window
Show Selection of the display mode for data. Activates either the graphics mode or the respective
selected SPL value in very large letters.
Update - Update the graphics choice of new parameters.
Default - Set the default values.
Image 7.2.5: SPL Statistics
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The main menu includes the following commands:
File
Save SPL history file - Save the SPL data as spl file..
Open SPL history file - Load of spl files..
Export ... - Exports data to text format
ASCII (1s Logged) - Exports Leq, SPL and Lpeak sections in seconds
ASCII (100ms Logged) - Exports SPL (Almost) in 100ms sections
CSV (1s Logged) - Exports Leq, SPL and Lpeak in seconds sections in CSV format
CSV (100ms logged) - Exports SPL (Almost) in 100ms sections in CSV format
File statistics and user info - SPL stats and user-entered information
current. spl file (see Figure 7.3.5). With copy the data is copied to the Clipboard.
Edit
Copy - Copy the graph to the clipboard
B / W background color - Switch to Bk / Wh
Setup
Calibrate audio device - Opens the Calibration
Setup audio devices - Opens the setup menu for the sound card
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7.3
Detection of resonances including downsampling
Resonances, whether space, housing or unwanted cone resonances are usually
Phenomena. Since their formation is not to prevent, to the sound harmful
Effects are only targeted minimized. This assumes, however, that at least the position
and the quality of the resonance are also known. In some cases, the simple
To obtain funds, in other cases a bit more effort is already required. Some
fundamental considerations in Detection of audible Resonances [24] run. Here
are only some examples of measurements are presented for entry.
Room resonances
The metrological detection of room resonances (modes) should ideally
be preceded by theoretical considerations. For rectangular rooms, the modes calculated
following formula:
f= Frequency of the mode in Hz, cSpeed of sound = 344 m / s at 21 ° C
nx = order of the mode space length, ny = order of the mode width of the room,
nz order of the fashion area height (nx, ny, nz = 0,1,2,3, ...)
L, W, H = Length, width and height of the room in meters
The following example of a space with dimensions L = 5.00 m, B = 3.90 m, H = 2.20 m
the spatial modes were calculated. Compare calculated and measured relative to the location
the room resonances (Figure 7.3.1).
34.2 Hz
102.9 Hz
44.0 Hz
103.8 Hz
55.8 Hz
111.5 Hz
68.6 Hz
111.9 Hz
78.0 Hz
112.8 Hz
81.5 Hz
117.5 Hz
85.2 Hz
87.9 Hz
122.4 Hz 129.1 Hz
89.5 Hz
95.8 Hz
131.9 Hz 135.4 Hz
7.3.1: Measurement of a speaker in the room (see also Section 6.0.2)
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Next determined if the position of the resonance, the quality or the duration of the Abklingvorganges
to be, this is possible using CSD or burst decay.
Image 7.3.2: CSD and burst decay for the determination of room resonances
Picture 7.3.2 shows the analysis shown in Figure 7.3.1 for example, at a sampling
(Sampling rate) of 48 kHz. In the burst the resonances decay is independent of the
To identify sampling frequency in the range <200 Hz good, in CSD, however, is more guesswork
announced.
8kHz
16kHz
32kHz
96kHz
Image 7.3.3: CSD with different sampling frequencies
The problem can be solved by reducing the sampling frequency. 7.3.3 image shows that the
low-frequency resolution with decreasing sampling frequency increases. For the 8 and 16 kHz
lowest decay modes with respect to location and to identify good.
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As of version 1.6.2 ARTA provides a down-sampling function. Allows PIR files
any resolution at lower sampling rates - for analysis of low-frequency
Room modes - be reduced. Sampling
rates from 4 kHz to 8 kHz should be good
Deliver results.
Download the desired file and pir activated
fours, then in the Impulsant-
word up in the "Edit" menu, select "Resample to lower frequency. " Now you can
the new sampling rate and anti-aliasing factor (cutoff frequency of the anti-aliasing filter,
see picture set 7.3.3a). In the specified
Range from 0.5 to 0.95 provide all factors
good results, the default value of 0.9 is
however, the recommendation of Ivo Mateljan.
After downsampling, the
Frequency response above fsampling / 2 = 4 kHz /
2
cut off (see Figure 7.3.3b, right
middle image).
Image 7.3.3a: effect of anti-aliasing factor (0.5 = black, red = 0.95)
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96 kHz
4 kHz
Image 7.3.3b: Comparison of PIR, FR and CSD before (left) / after (right) down-sampling rate of 4
kHz
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Speaker cabinet
Applies to speaker housing above said regarding the modes equally, because they provide so
nothing more than "small rooms" that only the frequencies shift to higher
Regions.
1m line, closed
1m line, open, empty
1m line, open, 1 Mat
Frequency
/ Phase
Impedance
Distortion
Burst
decay
CSD
Image 7.3.4: Resonanzdetektierung for different speaker cabinets
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Picture 7.3.4 shows measurements on a 1 m long transmission line in open (middle) and in the
closed state (left). In addition, a measurement of a light attenuation was
Line carried (right). All shown measurements (frequency, phase, impedance, distortion burst
Decay CSD) respond to resonances proves to be particularly sensitive to the impedance measurement.
The next example shows that a reevaluation Study material, by Thomas
Ahlers Meyer [23] was carried out. The complete results at the specified location
be studied in detail, here is only an evaluation of the WAV files with ARTA
shown. For mediation of an extract from the experimental program, first the
Impulse responses for the measured material combinations shown.
Image 7.3.5: Decay behavior of different material combinations [23]
Picture 7.3.5 shows the measurement range for 16mm MDF with different "coatings". In the lower
Range of partial images can be seen as a reference (green) each 16mm MDF board. Please
note the different scales in the overall comparison of the materials.
Picture 7.3.6 shows a different treatment of this measurement files (frequency response, burst Decay,
Burts
Decay sonogram). In the left photo series 16mm MDF is in red as a reference. In
Burst decay (middle) and in the sonogram (right) comes very well the effectiveness of the
different measures out.
This program is about to be continued with an accelerometer.
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16 mm MDF
+ 4mm
bitumen
+ 6mm tile
+ 10mm soft
fiber
+ Bitumen +
plywood
+ Glue +
plywood
Image 7.3.6: Decay behavior of different material combinations [23]
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Speaker
When the drivers are, the membrane resonances of particular interest. The classic
Method for detecting the impedance measurement. As already indicated above, this reaction
Method is very sensitive. To separate the diaphragm resonances of resonances, is
of course "free air" measured.
Image 7.3.7: Impedance response of a woofer
Picture 7.3.7 shows the impedance characteristics of a classic, the KEF B139. It is clearly seen,
that the membrane is between 700 Hz and 2 kHz resonance has problems. Is the frequency response
the visible. 7.3.8 Image shows measurements of different sensors (microphone (blue)
Accelerometer (red), laser (black)). Both the microphone and the
Accelerometer suitable for detecting diaphragm resonances.
Image 7.3.8: Frequency response (blue), diaphragm deflection (black) and acceleration (red)
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Image 7.3.9: Frequency Response bead (black), dust cap (red) and impedance (gray)
Picture 7.3.9 shows the results of two near-field measurements. The black curve in the midBead, the red center of the dust cap measured. In the region around 300 Hz, both curves start to
diverge, which is also visible in the impedance response in the form of an irregularity.
Such Erscheiningen may be caused by resonances in basket, dust cap or diaphragm
are.
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7.4
Create wav files to the external excitation signal with ARTA
Soon ....
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8
Dealing with data, data files, shortcuts, etc.
Everybody knows it: Quick check something. So measure, store or print data and
finished. Some time later, a fall of such printouts or data files to the hands and
large brood begins: How was it like? How was measuring distance? Could indeed from the
Be determined impulse response, but was not saved. What were the
Boundary conditions, as this was, how was that, and finally, why and in what
I then related the measurements carried out at all?
What does this tell us? Each measurement should be planned and documented. I.e. the aim and the
Purpose should be defined, it should be clear what the main influencing parameters are respectively
especially what to look for and it should first be considered as filed or documented
will. ARTA offers regarding documentation and traceability of measurements, a series
of functions on, but the only help if they are applied!
Basically, it is recommended that each of the respective measurement always Urformat (PIR, LIM,
Save HSW), because only in this format, all other evaluations obtained
are. If it is already evaluated directly during the measurements, it has been proven that
Results (eg graphics) to copy into a word processing document open in parallel and
immediately with comments.
8.1
Graphical representations in ARTA
ARTA has no direct printer output, but various options, charts or
Prepare or to format graphics for further use. The next two
Sections will provide a brief overview of the possibilities of ARTA.
8.1.1 Output and formatting charts
The output normal "screenshots" of the entire window is very easy, by the
Shortcut Ctrl + Print the image in the clipboard is stored and can then in each
Figure 8.1: Screenshot of a complete FR window
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open Windows application such as Word, PowerPoint, etc. are copied.
To obtain a copy of the "naked" chart in the window, we use either the
Shortcut Ctrl + C, the menu 'Edit> Copy' or the 'Copy' button in the current window.
In the main window of the 'Copy' button is an icon
shown.
The command opens - as shown below - the dialog 'Copy to Clipboard with
Extended Information ', provides the user with four options:
Figure 8.2: Copy-Menu for graphics output
1) In the input box at the top of the menu can be used to explain the measurement of any text
are inputted. He appears in issue directly below the graph.
2) 'Add filename and date ' activates the output of file name, date and time of the
Graphic.
3) 'Save Text ' stores the current input. It is the next time again
And can be modified as desired.
4) 'Select bitmap size' the size of the graphic is determined:
Current screen size
Smallest (400 pts)
Small (512 pts)
Medium (600 pts)
Large (800 pts)
Largest (1024 pts)
- Current size, width, and height are variable
- Pre-defined graphics with 400 points
- Pre-defined graphics with 512 points
- Pre-defined graphics with 600 points
- Pre-defined graphics with 800 points
- Pre-defined graphics with 1024 points
The options with a defined size have a fixed ratio of width to height of 3:2
on. With 'OK' image is copied to the clipboard, 'Cancel' aborts the operation.
When applying all the above possibilities to influence / addition of a graphic
see the screenshot of Figure 8.1 is displayed as shown in Figure 8.3. In the footer are now FileName,
To see the date and time and explanatory text. The text size is a maximum of 128 characters.
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Figure 8.3: Graphics with information about the file and explanatory text
8.1.2 Working with overlays
Overlays are temporarily stored, and a collapsible measuring curves. They facilitate the
Measuring life considerably, because so are direct comparisons of different variants of
e.g. Enclosures and crossovers possible. The opportunity to work with overlays, it is
all products the ARTA family. In the following the application of overlays is based
some examples are discussed.
The main application of overlays is in the frequency range (Smoothed Frequency
Response), but also in the time domain (impulse response) there are meaningful applications.
Figure 8.4: Smoothed Frequency Response - Windows - Overlay
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In the "Smoothed window frequency response " can the current curve or the targets for
Filters can be defined as an overlay. A further manipulation of overlays is in
Menu of the same name with the following options:
Set as overlay - Saves the current graph as an overlay
Set as cursor overlay Below - Stores the left part of the curve
the cursor as an overlay
Set as cursor overlay Above - Stores the part of the curve
right of the cursor as an overlay
Load overlays - Invites overlay file
Save overlays - Secures file as an overlay
Management overlays - Enabled 'FR Overlay Manager' for the
Editing of names
Delete last - Deletes the last overlay
Delete all - Deletes all the overlays
Generate target response - Generate targets for
Standard crossover
Load target response - Invites any target files as txt
Delete target response - Deletes the target for standard frequency
give way
Load impedance overlay - Impedance loads files (txt, zma, imp)
Preparation of the common frequency and
Impedance transitions
Delete impedance overlay - Deletes the impedance overlays
Any further processing of overlays can be used in the mask 'FR Overlay Manager ' (See
Figure 8.5) are made. It is the 'overlay command Management overlays'
opened.
Figure 8.5: Menu "FR Overlay Manager"
Some commands (Add, Add above crs, crs below Add, Delete all) We are already in the
known parent menu, the rest is explained below:
Replace sel - Replaced by current elected overlay curve
Delete sel - Deletes all selected overlays
Color - Changes color for highlighted overlay on the 'overlay menu Colors'.
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A click on the commands listed below to solve the following reactions:
Single click - Select desired item
Single click on check box - Overlay makes visible or invisible
Double - Enables editing of the overlay name
By pressing 'check All ' to all existing overlays are enabled.
Please note that the available space is limited below the graph. If you
e.g. have very long file names, so it is advisable to reduce this. Highlight
to the FR overlay manager with the cursor on the appropriate line and write the
existing text as desired (see example below).
Largest (1024): Full text
Smallest (400): Full text
Smallest (400): Reduced text
Figure 8.6: Customize the caption
As of version 1.4 are also overlays the impulse response
Window available. The corresponding menu located at the top
the main menu bar (see left).
The childrens' menu items and content differ
indistinguishable from those in the smoothed frequency response
window, they
are only slightly reduced, since only with impulse responses
being worked on. Therefore, at this point no further
Explanation.
The menu item "overlay Info " shows in the
left-image information displayed on charged
Overlay file.
Figure 8.7 shows the impulse responses of a midrange speaker (TMT = blue = current measurement) and
a tweeter (HT = red = overlay). In this illustration, the time lag between very good
to see the two chassis.
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Figure 8.7: Overlays in the time domain
Figure 8.9: Overlays in the time domain (left = zoomed in, right = bold line)
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8.2
Editing data and data files
ARTA provides some functions for documentation, editing or manipulation of
measured data. Access to the functions via three menus. It should be noted,
that the operation of almost identically worded commands in the time and frequency domain
is different.
Time range
New - Clears the memory
Invert - Invert the impulse response (see 8.1)
Open - Opens PIR data files
Save - PIR stores data files
Save as - PIR stores data files under
another name. Attention, ARTA overwrites
Files without question. If you summed
or have scaled the modified PIR File
Always Save with this command.
Rotate cursor at - Cuts the impulse response
before the starting cursor position.
Scale - Scale the impulse response by
any mathematical operations (see
Example)
Info - Plenty of space for comments on the
Measurement.
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Export - Export various data formats
Import - Import different data formats
Load and Sum - Summation of
Impulse responses (see 8.1)
Frequency range
Cut below / above cursor - the current
Frequency response is part of the left / right of the
Deleted cursor
Scale Level - Scaled frequency response
desired level
Subtract overlay - Subtract the overlay
the current frequency response
Subtract from overlay - Subtract the
current response from the overlay
Power average overlays - All existing
Overlays are averaged.
Merge overlay below / obove cursor - The
Overlay is left / right of the cursor with the
current curve associated
The upper part of Figure 8.2.1 shows the effect of "Cut below / above cursor ". In this
The case was cut off running to the left. In the "Time-Bandwidth
Requirement " is "cut below cursor" is used (lower half).
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Image 8.2.1: Function of "cut below cursor" and "Time-Bandwidth Requirement"
Picture 8.2.2 shows the measurement of a small wide-range speaker with two different microphones
(NTI M2210, T-bone MM-1). The NTI M2210 is a Class I microphone and is here for reference
used to generate a compensation file for the inexpensive measurement microphone MM-1. In
Image 8.2.3 is the actions of the functions "Subtract overlay" and "Subtract from overlay"
shown. In the arrangement shown here would be the "Subtract overlay file" as
Use the compensation function for the MM-1.
Image 8.2.2: Overlay = NTI M2210, T-bone measurement = MM-1
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Subtract overlay
Subtract from overlay
Image 8.2.3: Function of "Subtract overlay" and "Subtract from overlay"
Picture 8.2.4 shows the effect of "Power average overlay " using the measurements of the
Radiation from a small woofer and a tweeter in 10 ° increments.
The red curve shows the respective averaging over all overlays.
Image 8.2.4: "Power average overlays"
To use this function is the study of "Testing Loudspeakers - Which
Measurements Matter " Part 1 and Part 2 by Joseph D'Appolito,
http://www.audioxpress.com/magsdirx/ax/addenda/media/dappolito2959.pdf
http://www.audioxpress.com/magsdirx/ax/addenda/media/dappolito2960.pdf
as well as the highly recommended book by Floyd E. Toole, "Sound Reproduction Loudspeakers and Rooms " Elsevier in 2008 recommended.
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8.3
Scale and Scale Level
Below you will find a small collection of formulas with common computing and
Adjustment functions:
Scale
Scale Level
= D / dN
= 20 log (d / dN)
Nahfeldpegel PNF on far-field level
PFF adjust (half space, 2pi)
= (R / 2 d)
= 20 log (r / 2d)
Nahfeldpegel PNF on far-field level
PFF customize (Free, 4 Pi)
= (R / 4d)
= 20 log (r / 4d)
Level normalized to dN in the far field
Level adaptation reflex port to PP
Membrane PD in the near field
= (SP / SD) 0.5
= 20 log (SP / SD) 0.5
Legend
D
dN
SP
SD
R
PNF
PFF
= Measuring distance
= Reference distance (1m is the usual reference distance)
= Area of the reflex tunnel
= Area of the membrane
= Radius of the diaphragm
= Level of near field
= Level of far-field
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8.4
Keyboard Shortcuts - ARTA effective use
Although the mouse is the most commonly used tool at the desk, so it is not
always the most effective. So-called "Keyboard Shortcuts " shorten the sometimes cumbersome
Way through several menus from. ARTA provides some of these "shortcuts".
Key / key combination
Function
Up and Down
Ctrl + Up and Ctrl + Down
Left and Ctrl + Left
Right and Ctrl + Right
Left Shift + Left or Right
Right Shift + Left or Right
PgUp and PgDown
Ctrl + S
Ctrl + O
Ctrl + C
Ctrl + P
Ctrl + B
Shift + F12
2 x ALT + R
Changes the time shown on the screen gain
Changes the offset (the overlay is unaffected)
Shifts the graph to the left
Shifts the graph to the right
Moves the cursor left or right
Moves the marker to the left or right (if available)
Changes the zoom factor
Saves the current file
Opens a file
Copy a picture into the clipboard (user-defined)
Copies the current window to the clipboard
Changes the background color (Color / Black & White)
Evaluation of Farina sweeps (see section 7.1)
Repeated measurement with the same settings
ALT + M
ALT + P
ALT + G
Window shows the magnitude (frequency response)
Window shows the phases (phase transition)
Shows the group delay window
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9
Measure with STEPS
STEPS allows the measurement of the frequency response and harmonic distortion of loudspeakers
using a stepped sine wave (stepped sine). All major commands and operations for STEPS
are included in the top menu bar.
In addition to the above basic functionality offers
STEPS in the menu record the following special
measurements:
•THD vs.. Amplitude (see Section 9.3)
•X vs. linearity. Y (see Section 9.3))
•Membrane deflection vs.. THD (see Section 11.1)
Rules must be observed when working with STEPS, that the energy content of the
Is excitation signal (stepped sine) compared to the higher noise signals. To damage from
Test (DUT) and to prevent the equipment from the first measurement should
basically control the output level.
Since many symbols / controls are identical to those of ARTA, below is only
discussed the specifics of STEPS.
9.1
Basic setting of STEPS
In the menu 'Measurement setup' (See Figure 9.1) are all important parameters for measuring STEPS
adjusted. The menu is in the areas of system (Measurement System) and generator (Stepped
Sine generator) and a peak meter divided for setting levels.
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Figure 9.1: Measurement setup in STEPS
The fields or parameters contained in it have the following meaning:
Measurement System:
Measurement mode
Choice between one-and two-channel measurement (see
also section 9.2)
Response channel
Selecting the input channel (default = Left)
Sampling frequency
8000 Hz to 96000 Hz
Min integration time
(Ms)
STEPS determines the frequency response of the portion of the signal to
the "I / O Latency" and "Transient Time" begins by integrating the
Sine wave signal in the time domain. This time "integration time" is called.
The time required depends on the lowest frequency desired.
When a signal with the lowest frequency F [Hz] is to be measured,
then the minimum integration time 1 / F [s] must be. For a
Frequency of 20 Hz the integration time ie 1/20 = 0.05 s = 50
ms.
Furthermore, ARTA and STEPS use a special filtering of the
Signal by applying a windowing to Kaiser. This requires
minimum of 5 complete cycles (250 ms at 20 Hz). If you have a
wants faster measurement, which can only by increasing the lower
Frequency can be achieved.
Note: For Klirrmessungen should double the fundamental
Be used integration time (ie at 20 Hz 500 ms).
Transient time (ms)
The measurement of the sinusoidal signal is in steady state to
occur. When the steady state is reached depends on the
Resonance behavior of the system or from the acoustic reverberation.
For measurements in space should the "Transient Time" at least 1/5 of
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Amount reverberation time. Values between 100 ms and 200 ms for
usual normal rooms. For outdoor measurements, the "Transient should
Time "will be set to 50 ms to 100 ms.
I / O delay (ms)
Due to the distance between the microphone and speaker
always a delay exists. A reasonable phase response to
get, this delay must be considered.
Intra-burst-pause (ms)
After a measurement, the system must swing out only again before
can begin the next measurement. This break is
"Intra burst pause" called. As a rule of thumb for the duration of the "intra-burst
Break "can again be assumed 1/5 of the reverberation time.
Stepped Sine Generator:
Start frequency
Choice of start frequency in Hz
Stop frequency
Choosing the stop frequency in Hz
Generator level
Entering the generator output voltage in dB re FS
Frequency increment
Step size of the frequency steps (1/12, 1/24 or 1/48 octave)
Mute generator Switch-off transients
Checkbox is active = calculated elimination of clicking noises on
End of the signal incur. This will affect slightly longer measurement times.
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9.2
Amplitude frequency response and distortion measurements with STEPS
Frequency response measurements with STEPS run on principle as with ARTA. The main
Difference lies in the excitation signal (see original manual) and the duration of the measurement.
Depending on the choice of parameters can already take a few minutes to measure. From the sum
of 'integration time', Transient Timeline 'and' intra-burst break 'multiplied by the, frequency
Increment 'and the number of octaves swept resulting in about the time required. It
therefore recommended for initial trials to be cautious not only to the signal level,
but also with the frequency resolution (1/12 Oct).
The choices made when choosing the display amplitude, phase, amplitude and
Phase and amplitude distortion and THD in%.
To get a proper phase relationship between the speaker and the path must
Microphone can be compensated by a delay. It is difficult to determine the exact value
because for this we would have the exact location of the acoustic center of the speaker know (see
Section 6.3). In a reasonable approximation, we can calculate the delay as follows:
I / O delay [ms] = 1000 x measuring distance [m] / speed of sound [m / s] c = 344 m / s
For a measured distance of 0.5 m, the delay must therefore be 1.4534 ms.
Image 9.2.1: Frequency response of a 6 "TMT measured with 1/12 octave resolution with STEPS
As mentioned above, STEPS also provides the measurement in single and dual channel mode. In
Unlike ARTA but the current absolute level is shown in single channel mode and
no reference level.
If so increases the output voltage of the amplifier or in single channel mode with STEPS
is reduced, then displays the level in the frequency response. This is sometimes quite
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useful if you want to know at what level output voltage of the amplifier which
the microphone is applied (see Figure 9.2.2)
Dual Channel (-1dB to-12dB)
Single channel (-1dB to-12dB)
Image 9.2.2: Amplitude frequency response in one-and two-channel mode at STEPS
As in dual channel mode (Figure 9.2.2, left) shows the reference level (eg dB re
20uPa/2.83V), is any change in the output voltage - depending on the choice in the menu "View"
"Sound Pressure Units" - back-calculated from STEPS back to 2.83 or 1 volt. In
Single channel mode, any change in the output voltage of the amplifier in the level
displayed (Figure 9.2.2, right).
Besides the measurement of frequency responses STEPS is particularly suitable for measuring
Klirrfrequenzoutputs suitable. The measurements with STEPS are less prone to interference, but take - as already
noted above - depending on the setting for much longer than the Farina method (see Chapter
7.1)
The control of the display (dB or%) using the buttons in the top menu bar.
Magnitude = M + D + Distortion
D =%% Distortion
Image 9.2.3: Options to display Klirrdiagrammen
In Klirrmessungen Note that the result of both components of the electrode
can be influenced, as well as by the measurement environment. When to make long distance
measurements
the spatial influence in measurement results very noticeable and the comparability of
Results is given only to a limited extent.
Therefore, some attention is attached here. Careful experiments with different Perules and measurement intervals give an impression of the effect of various influencing
sizes. In order to exclude influences of space and increase the comparability of results
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Nisse should Klirrmessungen within the reverberation radius (see Section 6.2) or in the near field carried
will result.
If measured in the near field, it must be ensured that neither microphone nor Mikrofonvoramplifier can be overridden. Klirrmessungen for specifications - so the recommendation of AES21984 - should be carried out at about 10% of the rated power of the speaker. These are
not uncommon level of 90 dB or greater at one meter distance measurement. To a field measurement
transmitted (see Figure 6.2.8), a level of 120 dB would lie on the microphone. This Peyellow is rich for many inexpensive microphones have the limit of the maximum permissible
Sound pressure.
Here is an example of a microphone test that meets the conditions described above in about.
The test candidates were a very cheap microphone (MM-1 T-bone, about 35 €) and a
The microphone mid-region (Audix TM1, about 300 €) to the test. Served as a reference
a Class I measuring microphone (NTI M2210, about 1100 €).
Picture 9.2.4 shows the t-bone in direct comparison with the reference microphone (THD, D2, D3, D4).
It is clear that the T-bone is not necessarily Suitable for demanding Klirrmessungen,
because the deviations from the reference microphone are significant.
THD: T-bone (gray)
D2: T-bone (blue)
D3: T-bone (red)
D4: T-bone (green)
Image 9.2.4: Direct Comparison of T-Bone MM-1 vs. NTI M2210
The second test candidate reflected in this discipline, however, more than decent (see Figure
9.2.5). The Audix TM1 and the reference microphone hardly differ.
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THD: Audix (gray)
D2: Audix (blue)
D3: Audix (red)
D4 Audix (green)
Image 9.2.5: Direct comparison Audix TM1 vs. NTI M2210
This comparison will u.å. show that there is a dispute about decimal places Verzerrungsmessolutions with cheap measuring equipment is not worthwhile.
As mentioned above, playing alongside the equipment used and the boundary
conditions under which the measurements are performed an important role. Also, this is now a
some measured examples. Picture 9.2.6 shows Klirrmessungen a 5 "TMT in the near field and in 10, 25
and 40 cm measured in dB and%.
With increasing measurement distance, the space makes both in frequency response and in the
Klirrverlauf noticeable. Picture 9.2.7 shows a direct comparison of the Klirrfrequenzganges in%
(THD, D2, D3, D4) and in the near field 40 cm. In addition to increasing "unrest" in the course adds to the
Space also Klirranteile added.
The examples show that for producing reproducible Klirrmessungen
is something to consider. As mentioned above, it is in compliance with excitation level,
SPL the microphone, measurement distance and the suppression of interfering reflections
To search for each measurement arrangement the current best compromise.
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Image 9.2.6: STEPS Klirrfrequenzgang in the near field and in 10, 25, 40cm distance (from above)
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THD: FF (red), NF (blue)
D2: FF (dark blue), LF (light blue)
D3: FF (red), AF (Purple)
D4: FF (green), NF (brown)
Image 9.2.7:
STEPS Klirrfrequenzgang in percent. Comparison THD, D2, D3 and D4 in the
Near field and 40 cm working distance.
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9.3
Voltage or power-related measurements with STEPS
STEPS includes the features already described in section 9.2 in the Record menu three
other special functions:
•Distortion vs.. Amplitude
•Linearity Function
•Loudspeaker Displacement / Distortion
The third most mentioned function is described in detail in Application Note No. 7 [VII], the
first two are explained below with reference to some application examples.
With the "Distortion vs. amplitude " Function to voltage or power-related
Distortions in both electrical (eg amplifier) and electro-acoustic systems (eg
Speaker) are measured, wherein the performance-related values on the respective
Reference resistance are converted (P = U2 / R) must be handwritten and the X-axis
may be added.
Image 9.3.1: Distortion vs. amplitude menu
In the left part of the screen (Figure 9.3.1) are the input fields for the measurement parameters, the
lower
Edge of the picture, the setting parameters for the graphics and overlays.
In the "General Distortion Measurement " , in addition to
Input channel and sampling different evaluation modes
be selected (THD, IMD DIN, CCIF IMD). For details, see
In the original manual of STEPS
In the "Excitation Sine Voltage Range " the measured parameter set (frequency,
Start and stop value, linear or logarithmic increase in the voltage, the number of stages). Below
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the stop value is the maximum. Output voltage in V appears. This is calculated from the
Gain of the power amplifier and a safety margin to the full scale
of 3 dB (see also section 3.2). Attention, please think before starting the measurement,
if your test object may be damaged at full!
Figure 9.3.2 shows the Klirrverlauf a small power amplifier at 1 kHz as a function of
the voltage (power). Notes on the measurement and the measurement setup are described in Section 5.4
given.
Figure 9.3.2: THD vs. voltage @ 1kHz for a small power amplifier
Image 9.3.3: THD vs. voltage for different frequencies (left). THD at 3 volts (right)
9.3.3 Klirrverlauf image showing the (THD) at different frequencies depending on the
Voltage (left panel) for a 5 "midrange driver. The right-hand image shows the Klirrverlauf
in response to stroking rate of about 3 volts amplitude. The at 3 volts in the right image in
the designated measurement frequencies readings should recover in the right image
be.
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Image 9.3.4: Distortions menu (vs. deflection. THD)
Picture 9.3.4 shows the "Loudspeaker Displacement / distortion" or "Klippel Light" menu. The
full functionality is described in Application Note No. 7. At this point only to a
"Security feature", which is not in the "Distortion vs.. Amplitude "menu is included, presented
are. With reference to the above given information security is the termination function "THD
Breakfast Value " particularly interesting. Here, then, is not always the voltage to
ramped bitter end, but canceled due to "break THD value." Image
9.3.5 Klirrverlauf for showing the function of the voltage as a tweeter for two
different crossovers (18dB, 6dB slope) at 2.6 kHz. Here, the
Tweeter held braver than expected, so was the break value of 1% is not reached.
18 dB XO
Image 9.3.5:
6 dB XO
Klirrverlauf as a function of voltage for a tweeter with two
different crossovers (f = 2fs)
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In the second special feature of the "Linearity Function " , the relationship between
two variables are measured. In the menu section "Measurement Channels" are the
Possibilities are shown. It can be both the left and right channels or as excitation
be defined as recording.
Image 9.3.6: Linearity Function Menu (X vs.Y)
Picture 9.3.7 shows a simple linearity test with a cheap onboard sound card.
Image 9.3.7: Linearity test an onboard sound card
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10th Measuring with LIMP
LIMP allows the measurement of the impedance of the Ganges and the Thiele-Small parameters (TSP)
of
Speakers and systems. Furthermore LIMP was designed as LCR meters. As an excitation signal
There is a noise signal (pink-PN) and stepped sine wave (stepped sine) are available. The
Noise signal differs substantially from the energy content in the stepped sine. In this respect
it is not surprising if the TSP differ for different excitation (see Figure 10.6).
As with all programs of the ARTA family, are also LIMP the essential commands and
Operations in the top menu bar contains.
Figure 10.1: Opening screen of LIMP
10.1 Default of LIMP
In the following descriptions, it is assumed that for the ARTA MessBox
Available. As a reference resistor is a 27 ohm power resistor. Otherwise, an
the circuits for impedance measurements from Chapter 2 to build.
Before the first measurement, some basic settings are again made. The calibration of the
Sound card has already been discussed in Chapter 5, so here are only the special
treated LIMP.
The measurement setup of LIMP has three areas:
•
•
•
Measurement Config
Stepped sine mode
FFT mode
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In the area of measurement, the general config
Measuring parameter:
•Reference Channel: Default is the right input channel
•Reference Resistance: The ARTA is the MessBox
Specified with reference resistor 27 ohms. The
exact value is to be determined by measurement.
•Upper frequency limit: see below
•Lower frequency limit: see below
The frequency limits are also on the top menu bar
to control
In the area stepped sine mode, the parameters are
the stepped-sine excitation defined.
All parameters of this area are discussed in Section
9.1 explained. The default values are set for
conventional impedance measurements sufficiently.
In the field mode, the parameters for the FFT are
Defined excitation with pink noise.
•FFT Size: Number of values for the FFT (Resolution)
•Averaging: type of averaging (none, linear,
exponential)
•Averages max: maximum number of
Averaging
•Asynchronous Averaging: asynchronous averaging
on / off
Before it goes to the fairs, is to check whether the set output level not
Overdriving the input channels leads. It should be noted that the two excitation signals
are very different, i.e. when changing the level of excitation should be re-examined.
In the Setup menu generator all necessary settings are possible.
Generator
•Of excitation PN or Stepped
Sine
•Output level: 0 to-15dB
•Frequency for sinusoidal excitation
•Corner frequency excitation with
Noise signal
Input Level Monitor
Pressing the test is
signal started and set up
displayed in the peak level meter. Should
the indicator is red or yellow, the
Level to reduce.
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10.2 Determination of the TSP
LIMP supports the determination of TSP in two different ways:
•
•
in the test case of a known volume (volume method, image 10.a)
Complaining through the membrane with a known added mass (mass method, Figure 10.2b)
In principle, both methods are equivalent, but the volume method should always
be used if the resonance frequency of the speaker is very low. When
Mass the resonant frequency method could then be reduced until the linear
Measuring range of the sound card will be left.
Image 10.2a: Volume Method
Image 10.2b: Mass Method
10.2.1 Volume Method
The procedure for determining the Thiele-Small parameters for the volumetric method is as
follows:
Set 1) test volume.
Is a function of the diameter (membrane surface area) of the speaker to be measured
Test to obtain housing with appropriate and known volume. A rough estimate of the
volume required for the test case can be made with the following table
are. However, it need not be specially manufactured for each
known housings of this size is suitable provided
It leads to a shift of resonance of the speaker in the field of
between 20% and 50%. The resonance shift is by LIMP
controlled and displayed (see Figure 10.5a).
Example: According to the table is for a 17 inch chassis, the volume of
Test case are approximately 12 liters. When entering the
Test volume (Closed Box volume) in the "Closed Box Method"
it should be noted that, at the exact volume of the test box, the volume of
Speaker cone must be added (see Figure 10.2a).
2) Calibration
Move the switch SW1 to position the ARTA MessBox, Impedance Measurement 'and SW2
Position Imp Cal. With CAL from the top menu bar, the menu, Calibrate Input Channels'
Open and calibrate the system (Figure 10.2c).
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Image 10.2c: Calibration menu in LIMP
3) Measurement of the speaker "free air"
Figure 10.3: Impedance response of the LS without
housing
3) The free-air measurements with Overlay
Set save (yellow curve)
4) The measurement of the LS in the test case procedure (see Figure
10.2a)
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Figure 10.4: Impedance response of the LS in the housing (black) and free air
(red)
5) Using the menu Analyze Loudspeaker parameters - Closed box method can now
required parameters (RDC, DD, VT) in the area, users Input ' are entered (see also
10.2.1).
Note: If you do not need a full set TSP, but only EBP = (fS / QES)
want to specify, press to measure the impedance response of the LS without housing
, Calculate '.
Image 10.5a: Menu to calculate the TSP
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If the fields are closed for entries (gray shading), is still no overlay
defined. By pressing, Calculate 'LIMP calculated the TSP (see Figure 10.5b).
Image 10.5b: Calculated TSP
6) Copy
By 'Copy', the calculated values are copied as ASCII to arbitrary files. The
Output appears as follows:
Thiele-Small parameters:
Fs = 79.85 Hz
Re = 5.75 ohms [dc]
Qt = 0.63
Qes = 0.68
Qms = 8:02
Mms = grams 13:47
Rms = 0.842902 kg / s
Cms = 0.000295 m / N
Vas = 6.64 liters
SD = 126.68 cm2
Bl = 7.555168 Tm
ETA = 0.48% to
Lp (2.83V/1m) = 90.33 dB
Closed box method:
Box volume = 5.40 lit
Diameter = 12.70 cm
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10.2.2 Mass Method
The procedure for determining the Thiele-Small parameters for the mass method is as follows:
1) Set test mass.
Is a function of the diameter (membrane surface area) of the speaker to be measured
Test mass to raise with the appropriate weight. Also in mass method (Fig. 10.2b) is
by applying an additional mass
Resonance shift between 20% and 50%
be achieved. An additional mass in the
Size of the membrane material into MMD
as to a reduction in the resonant frequency of
30%.
If the membrane mass is not known, so
can in the diagram opposite a rough
Assessment to be made.
Example: For an 8 inch chassis MMD acc.
Diagram between 15 and 50 grams respectively.
An additional mass of 25 grams should the
first attempt to adapt and evaluable
Measurement result.
2) and 3)
Points 2 and 3 are handled analogously to the volume method.
4) The measurement of the LS with additional mass to perform (see Figure
10.2b)
Figure 10.6: Speaker (red) and without (green) additional mass
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5) Using the menu Analyze Loudspeaker parameters - Added Mass Method can now
required parameters (RDC, DD, MADD) in the field, users Input ' are entered (see also
10.2.1).
Figure 10.7: Input and calculation of the TSP
6) Copy
Through, copy to Clipboard ' or "export to CSV File " to the calculated parameters
be exported. If several measurements with statistics is to be operated, recommended
the CSV export, because then the full functionality of Excel.
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10.2.3 Notes on TSP measurements
Let us in this section Duch, the Master Class Seminar "Loudspeaker Parameters" of
Neville Thiele and Richard Small from 2008 forward [31]. There were - among other
Issues - the conditions for the determination of TSP parameters (signal strength measurement location
(Horizontal, vertical), installation or restraint conditions) are discussed.
Strength of the excitation signal
Remember that you measure "small signal parameters"! But what is a small signal? Some
Standards and test specifications recommend measurements at one watt, ie U = √ P · Z = 2.83 V
at 8 ohms. This will allow smaller speakers already outside their linear region
be operated. Small recommends limiting the level of the measurement signal so that no straight
still a clean measuring signal is received.
In this context, it should be noted that the energy content in comparison to the LIMP
Bundling stationary measurement signals (PN, stepped sine) is different. Figure 10.8 shows impedance
gradients for both excitation signals with identical gain. The red curve is for the excitation
tion with, stepped sine 'and the black curve for the noise signal. It is clearly seen,
that the higher-energy sine signal as expected leads to a lower resonant frequency.
Figure 10.8: Impedance transitions at different excitation (red = stepped sine, black = PN)
It is well known that the resonant frequency as a function of the excitation
amplitude changed (CMS is not a constant). So far should not be expected that, under
different excitation strengths or even measurement methods provide identical results. It is important
however, the amount obtained under the particular conditions of parameter set is coherent
and therefore suitable for the calculation / simulation.
Clamping conditions
Clamping conditions under which the TSP should be measured? Firmly clamped, loose in
hand? Richard Small recommends a consistent holding in installation position (see Figure 10.11). He
showed in his presentation example of what happens when a mass-spring system unconsciously
more elements are added.
Figure 10.10 shows a trailing measurement example, in the chassis on different documents
(= Hard red, foam = blue) were measured. The chassis on the left of this picture was
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Membrane mass of 11g and 43g in the right part of image. It is very nice to see that
by the "activation" of soft, springy support an additional resonance is produced.
Picture 10:09: Influence of support for two speakers (red = foam, blue = MDF)
If it is already measured in a horizontal position, then you should at least a solid backing in
be selected. It should of course be at lying position measurement taken that a
any existing pole piece can breathe freely.
Figure 10.10: Measured position in the measurement
of TSP
With respect to the effect of the measurement position (horizontal, vertical) to the TSP of the chassis there
are
Some discussion forums.
Richard Small and others
Authors recommend that the chassis
to measure in the installation position, ie with
horizontal loudspeaker axis
[31], [27].
Should at least insofar
those who are permanently
the hobby loudspeaker
want to prescribe the construction
a simple holding device
reconsider, in which both the
"Normal mounting position" and
a fixed, non-compliant
Fastening can be realized. Since
Figure 10.11: jigs for TSP measurement
also the laws of physics
apply (action = reaction), is to fulfill the latter requirement a little mass
required.
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In ARTA Hardware & Tools Manual, but also in relevant forums, you will find this variety
more or less just nachzubauende examples. Relatively easy and inexpensive to
Implementation are solutions with straps (see Figure 10.11, left)
10.2.4 Measurement of the DC resistance with a cheap multimeter
If you are not sure if your multimeter to measure the DC resistance RDC
is accurate enough, you can make do with the following trick. Even with a very
simple multimeter can obtain sufficiently high accuracies with this method:
•Kill a known resistor RV (eg 8.2 Ohm ¼ Watt, 1%) and the speaker
in row
•Clamp to a 1.5 V battery
•Measure with a multimeter, the voltage across the resistor RV URV and the
Voltage ULS over the loudspeaker
•The DC resistance of the speaker is calculated as follows RDC = RV · ULS / URV
Example:
4 ohm woofer
Selected: RV = 4.7 ohms,
Measured: URV = 0.8368 V 0.5591 V = ULS
Calculated: RDC = 4.7 * 0.5591 / 0.8368 = 3.14 ohms (manufacturer = 3:10 Ohms)
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10.3 RLC measurement with LIMP
LIMP determines the value of resistors, capacitors and coils by calculating the
resistive, inductive or capacitive component of the impedance. Image 10.3.1 shows an example of
Impedance curve of a coil with a nominal value of 1.5 mH.
10.3.1 Image: impedance curve of a 1.5mH coil
On the menu Analysis RLC
Impedance value at cursor position receives
If the result is shown on the left.
LIMP points out that the measured
Impedance at the cursor position a
proportion of 0.776987 ohms resistive and
an imaginary part with an inductive
Value of 1.589mH
has. In the same way also pure capacitors or resistors are measured with LIMP.
For performing RLC measurements, it is important that, before the measurement, a calibration
is carried out, preferably is connected to the specimen.
Why calibration is required? Even with small differences in the sensitivity of
both input channels of the sound card (eg 0.1 dB), it may happen that under certain LIMP
Use delivers erroneous results since the phase close to 90 degrees, and an inductance
the one capacitor is almost -90 degrees.
In the event that the measured via the generator, and the voltage V1 across the impedance Z
measured voltage V2 due to differences in sensitivity of the two measurement channels
is corrupted, which can cause the measured impedance and phase values of more than
90 degrees and identifies the phase by 180 degrees makes a jump (see Figure 10.3.3).
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Image 10.3.2: Measurement setup for impedance
measurement
10.3.3 image showing the result of capacitance measurement without calibration. Up to 1200 Hz runs
the phase at almost +90 degrees and then conveys the impression that it is an inductance
concerns. 10.3.4 image showing the result of measurement by a calibration procedure. It can be seen that
the phase in the whole frequency range now behaves as expected.
10.3.3 Image: sRGB estimated impedance of a capacitor with 4.7uF/250V
10.3.4 Image: Calibrated estimated impedance of a capacitor with 4.7uF/250V
Not all LIMP users will have the problem described above. As stated above,
the problem exists only when the voltage V2 across the impedance is higher than the voltage V1
on the generator. To get around this, either the sensitivity of the probe
(Sample) can be changed or the input channels can be easily replaced. If the
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Input channels to be replaced, of course, must also reference the channel in the 'LIMP Measurement
Setup 'are exchanged.
Note:
In order to obtain correct readings for capacitance and inductance, the cursor should be on a
Frequency can be set, wherein the impedance is less than 100 ohms. This ensures that the
Measurements in the range of about 1% tolerance. The rationale is that the
measured impedance one of said sensors (sample) and the input impedance of
Sound card with typically 10k ohms are connected in parallel.
10.4 The accuracy of the impedance measurement
If clean test setup of the measurement error in the impedance measurement with LIMP less than 1%
should
be. If not given, it is often one of the following sources of error for this
responsible:
1 The sensitivity of the input channels of the sound card is different
2 The sound card has a too low input impedance (10-20 ohms)
3 The measuring cable between the power amplifier and speaker is too long
This can be remedied by the following measures:
1 Calibration of the sound card (see Section 10.2, Figure 10.2b)
2 A sound card with high input impedance using check (specification,
professional sound cards have input impedance of 1 MOhm) or a
Upstream input buffer.
3 For long measurement cables, the inductive or capacitive cable shares in the measurements
a. This also applies to transfer resistance at terminals or connectors. The motto
ie:
Short measurement using a cable with sufficient cross-section (1.5 mm2 or more)
b. If longer measuring cable must be used as the reference resistor
Attach near the speaker terminal. See also "a easier
Measurement setup for impedance measurement ... "
c. Ensure clean contacts. Only connectors and terminals of undoubted quality
use note).
Note: "Test leads" as shown in the right image are
often a source of error in measurements on loudspeakers. Since the cable
often are only clamped to the alligator clips are available in the
Sequence "variable contact resistance," which the
Reproducibility of measurements is not exactly conducive. When
You use such products, please check each connection
and possibly resolder.
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11th Recommendations for speakers specifications
The measurement of loudspeakers is not new, therefore it is not surprising that
the area is recognized normative. At this place - without limitation - two
Standards are:
-
AES2-1984 (R2003): AES Recommended Practice Specification of Loudspeaker Components
Used in Professional Audio and Sound Reinforcement
IEC 60268-5: Sound system equipment - Part 5: Loudspeakers
Standards are not always boring, they usually provide state of the art and should
In addition, concentrated knowledge or experience collections from the industry practice.
Below is an example of the list of requirements to a specification for bass and
Tweeter shown from the AES2.
Low Frequency Drivers
High Frequency Drivers
1
2
3
4
5
6
7
1
2
3
4
5
6
Dimensions and weight
Dimensioned line drawings
Mounting information
List of accessories
Description of electrical connections
Additional descriptive information
Physical constants, piston diameter, moving
mass, voice-coil winding depth and length, topplate thickness at voice coil, minimum
impedance Zmin, and transduction coefficient.
Thiele-Small parameters: ƒS, QTS, η0, VAS,
8 QES, QMS, RE, SD
Large-signal parameters: PE (max), X max, VD
9 Frequency response (0 °, 45 °) in standard
10th baffle *
Distortion (second and third harmonic), swept,
at 10% rated power
11th Impedance response, free air
Power handling in free air, 2 h
12th Displacement limit **
13th Thermal rise after power test
14th Recommended enclosures
15th
16th
Dimensions and weight
Dimensioned line drawing
List of accessories
Description of electrical connections
Additional descriptive information
Description of diaphragm and diaphragm
construction
Frequency response on plane-wave tube
7 (PWT ***)
Distortion on PWT; swept second and third
8 harmonics at 10% of rated power.
Impedance on PWT; swept
9 Dc voice-coil resistance
10th Power handling on Appropriate acoustic load
11th Displacement limit of diaphragm
12th Thermal rise after power test
13th
Comments:
* For the dimensions of standard baffle see Figure 11.2
** This recommendation has since been extended (see section 11.1)
Plane Wave Tube *** To see AES 1id-1991
Manufacturers should follow the recommendations in the data sheets of the AES. In serious
Manufacturers is also given, as a rule, data sheets of unnamed products should better
according to the motto: "Trust is good, control is better" to be treated. With ARTA, STEPS
and LIMP is the problem.
The Dimensions and mounting conditions (1-6) are usually at each
Manufacturers available, otherwise just remove the speakers. The figures below
Point 7 is one - if you do not want to check destructive - partially based on manufacturer
instructed other data can even be determined. The measurement of the Thiele-Small
Parameters (item 8, 12) is the domain of LIMP and is described in Section 10.2. With the
at 9 and 14 above large-signal parameters Xmax deals section 11.1 or
Application Note # 7. The measurement of the frequency response - even at angles - is the
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Field of activity of ARTA (Chapter 6, Application Note No. 6). Information for Standard Baffle
See the end of this section (Figure 11.2).
Item 11 can be used both with ARTA (Farina method, Section 7.1) and with STEPS (Chapter
9) do. Relative to the data max. electrical load (point 13, 15) one is in the
Usually rely on manufacturer data (2 hours continuous level could peace in the house or the
good neighbors seriously disturb).
Figure 11.1: Data sheet of a midrange speaker (Visaton AL130 - 8 Ohms)
Figure 11.1 shows the data sheet of a midrange speaker from Visaton. With the exception of item 11
and a number of parameters which can be calculated from available data, all
contain the information required. Instead of the frequency response at 45 ° is even a polar diagram for
attached representative frequencies.
Figure 11.2: Dimensions of the standard IEC baffle
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11.1 Determination of deflection XMAX
The AES 2 leads to the determination of the linear displacement of the
following:
Excursion of the voice coil (peak), wherein the "linearity" of the actuator differs by 10%.
The linear deflection can XMAX as% distortion of the input current or as%
Difference from the deflection of the input signal to be measured. The manufacturer
must appoint the method used. Further to the displacement volume (SD = VDPeak *
XMAX) can be specified.
This recommendation has been extended by AES initiative by W. Klippel and is now in the
Pre-standard "IEC PAS 62458: Sound system equipment - electroacoustical transducers Measurement of large signal parameters "flowed.
In Application Note AN4 [11] for the Klippel analyzer system is a procedure for
Determination of XMAX described in the following by way of example with ARTA
is transposed with:
1 Measure the resonant frequency fs of the speaker to LIMP. Choose this "stepped sine" as
Excitation signal. In this example, the resonance frequency was fs = 43.58 Hz determined.
2 Burdening the speakers in free field conditions with a two-tone signal with
f1 = fs = 43.58 Hz and f2 = 8.5 fs = 370.43 Hz and an amplitude ratio of U1 =
4 * U2 (see Figure 11.2) and run a series of measurements under variation of the amplitude of USTART
to
UENDE through.
Figure 11.2: Setting the measurement parameters in the "Signal Generator
Setup"
3 Measure the sound pressure in the near field and perform a spectral analysis to measure the amplitude
P (f1) and P (f2) and the harmonic components P (k * f1) with k = 2, 3, ... And K
Sum component P (f 2 + (n-1) * f1) and the difference component P (f 2 (n-1) * f1) with n = 2, 3
on the amplitude U1 through.
4 Measure the peak displacement X (f1) on the amplitude U1. A simple method for
Determining the deflection is as follows. Using a caliper with depth gauge
the distance to the dust cap is first measured, and the value with no signal as
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Zero recorded. Then the speaker is in the ARTA SPAMode with a
Sinusoidal signal excited at fs and the depth gauge carefully towards the dome
is pushed to a contact noise to hear. The relevant for the excitation voltage
determined value to be subtracted from the zero point to the corresponding deflection to
get.
5 Determine THD with the ARTA SPAMode at the resonant frequency with a sinusoidal excitation
Function of the amplitude U1:
Switch on "Two Sine excitation" and choose a frequency range between f2 + / - 2.5 * fs
linear representation.
Go with the cursor on the marked in Figure 11.4 and note the frequencies
respective level values. The second order distortion
and the third order intermodulation distortion
Figure 11.4: Determination of second and third order distortion
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calculated as indicated in the above formulas. It should be noted that the
read level values before inserting into the formula in absolute values are converted
(Para = 10 ^ (dB/20) need. The following table shows an example of the calculation
F
F143, 58
F2370, 4
f2-F1326, 9
f2 + F1414
f2-2f1 283.3
f2 +2 f1 457.6
P in dB
P abs
-48.6
-89.46
-87.95
-86.24
-103.63
0.003715
0.000034
0.000040
0.000049
0.000007
From these values, the second and third order distortion calculate
with d2 = d3 = 1.98% and 1.49%.
6 Find the minimum value in the range between U and USTART UENDE, wherein either
dt harmonic distortion or intermodulation distortion second or third
Order d2, d3 reach the 10% mark (U10%).
7 Determine the deflection XMAX for the corresponding amplitude U10%.
Figure 11.5: Determination of linear displacement according to [11]
Figure 11-5 shows the result of such a measurement procedure. In this example, achieved as a THD
First, the 10% mark and is thus a criterion for the determination of XMAX = 3.4 mm (see green
Arrows).
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Note: As of version 1.4, the method
automated. A detailed description
See the ARTA Application Note No. 7
[VII].
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12th ARTA Application Notes
[I]
No. 1: ARTA MessBox
[II]
No. 2: RLC measurement with LIMP
[III]
No. 3: Why 64-bit Processing
[IV]
[V]
No. 4: Determination of the free-field frequency
response
No. 5: The ARTA Mikrofonkalibrierkammer for lower end
[VI]
No. 6: Directivity and Polar
[VII]
No. 7: Determination of linear displacement with STEPS
[VIII] No. 8: In-situ measurement for estimation of absorption coefficients with ARTA
(Coming soon)
13th Literature
[1]
Mateljan, Ivo, "ARTA Manuals"
[2]
D'Appolito, Joseph, "Speaker Measurement", Elektor Electronics Publishing,
1999.
Struck, Temme, "Simulated Free Field Measurements" JAES, vol 42, no. 6, June 1994.
[3]
[4]
Dickason, Vance: "The Loudspeaker Design Cookbook", 4th Edition, Audio Amateur Press,
1991
[5]
Fasold and Veres: "Sound insulation and room acoustics in Practice", publisher of Construction,
Berlin, 1998.
[6]
Khenkin, Alex: "How Earthworks Microphones Measures"
[7]
AES2-1984 (R2003): AES Recommended Practice Specification of Loudspeaker
Components Used in Professional Audio and Sound Reinforcement
http://users.skynet.be/william-audio/pdf/aes2-1984-r2003.pdf
[8]
IEC 60268-5: Sound system equipment - Part 5: Loudspeakers
[9]
AN 4 - Measurement of Peak Displacement Xmax - Application Note to the
KLIPPEL ANALYZER SYSTEM (www.klippel.de)
[10] Griesinger, D.: Beyond MLS - Occupied Hall Measurement with FFT Techniques, 101st
Convention of the Audio Engineering Society, November 8 to 11, 1996, Preprint 4403
[11] Farina, A.: Simultaneous Measurement of Impulse Response and Distortion with a Swept
Sine Technique, 108 AES Convention, Paris, 2000.
[12] S. Müller, P. Massarani: Transfer Function Measurement with Sweeps, JAES, June, 2001.
[13] I. Mateljan, Ugrinovic K.: The Comparison of Measuring Room Impulse Response
Systems, Proceedings of the First Congress of Alps Adria Acoustics Association, Portoroz,
Slovenia, 2003, ISBN 961-6238-73-6
[14] D. Ralph: Measurement Techniques for speaker crossover design,
http://www.purespeakers.com/offsets.html
[15] D. Ralph: Finding Relative Acoustic Offset empirically, Speaker Builder 1/2000
[16] J. Kreskovsky: It's Just A Phase I Am Going Through,
http://www.geocities.com/kreskovs/Phase-B.html
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[17] John Vanderkooy: The Acoustic Centre
http://www.aes.org/sections/uk/meetings/AESUK_lecture_0604.pdf
[18] S. Fuhs, R. Höldrich, G. Tomberger: Validation of the law of distance and correction of the
Group delay and the acoustic center of the loudspeaker Adrienne process
[19] Thomas Ahlers Meyer: Acoustic "best" material for speaker cabinets
http://www.picosound.de
[20] Mateljan, H. Weber, A. Doric: Audible Detection of Resonances, Proceedings of the Third
congress of Alps Adria Acoustics Association, Graz, Austria, 2007
[21] Jerry Freeman, Techniques to Enhance op amp signal integrity in low-level sensor
applications Part 1 - 4), Planet Analog
[22] Mark Sanfilipo: Subwoofer Measurement Tactics (www.audioholics.com)
[23] ARTA Hardware & Tools Manual (coming soon)
[24] Measurement conditions Visaton (www.visaton.de)
[25] Mark Gander: Ground Plane Acoustic Measurement of Loudspeaker Systems, JAES Volume
30 Issue 10 pp.. 723-731, October 1982
[26] Melon, Long Run, Rousseau, Duke: Comparison of Four Measurement subwoofer
Techniques, JAES Volume 55 Issue 12 pp.. 1077-1091, December 2007
[27] Anderson: Derivation of Moving Coil Loudspeaker parameter using Plane Wave Tube
Techniques, Master Thesis, 2003
[28] Withhold, Waldman: Non-Linear Least Squares Estimation of Thiele-Small parameters from
Impedance Measurements, 1993, Preprint 3511
[29] J. Backman, Transducers Models
Handbook of Signal Processing in Acoustics, Springer 2008
[30] Mateljan, K. Ugrinovic: The Comparison of Room Impulse Response Measuring Systems,
Proceedings of the First Congress of Alps Adria Acoustics Association, Portoroz, Slovenia,
2003
[31] Neville Thiele, Richard Small: Loudspeaker parameter
Tutorial, AES 124th Convention, 2008
[32] Floyd E. Toole: Sound Reproduction - Loudspeakers and Rooms, Elsevier 2008
[33] Marcel Müller: Technical characteristics of sound card in the PC, 2005
http://www.maazl.de/hardware/sound/index.html
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14th Small Formula and images collection
Vrms
Vs.
Vpp
Vrms
1.4140 Vrms
2.8280 Vrms
Us
Us 0.7071
2,0000 Us
Uss
Uss 0.3535
0,5000 Us
-
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Series and parallel connection of chassis
In the series or parallel connection of a plurality of chassis of the same type, there is often the
Question of the resulting parameters. Generally chassis can both electrically in series
(Ser) and parallel (par) are switched. You can also acoustically in series
(Compound body) or are parallel (side by side in the box) is connected. For unique
Statements both can therefore required. Possible combinations and their
Effect on the parameters shown in the following table
[Http://cfuttrup.limewebs.com/].
1 LS
Electrically
Acoustically
fs [Hz]
Re [Ohm]
SD [cm2]
Mms [g]
Cms [mm / N]
VAS [ltr]
Rms [Ns / m]
WxL [Tm]
Le [mH]
Sqm
Qe
Qt
SPL [dB / V]
SPL [dB / W]
1
1
1
1
1
1
1
1
1
1
1
1
1
1
2 LS
Ser
Par
1
2
2
2
0.5
2
2
2
2
1
1
1
0
3
2 LS
Par
Par
1
0.5
2
2
0.5
2
2
1
0.5
1
1
1
6
3
2 LS
Ser
Ser
1
2
1
2
0.5
0.5
2
2
2
1
1
1
-6
-3
2 LS
Par
Ser
1
0.5
1
2
0.5
0.5
2
1
0.5
1
1
1
0
-3
..... more soon.
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15th Index
A
L
Acoustic fields
Far field ................................................. . 88
Free field ................................................. .. 88
Hall field ................................................. . 88
Near field ................................................. .. 88
Acoustic measurements
IEC baffle ................................... 204,
Speaker
parallel ................................................. 212
Series ................................................. .... 212
Specification ........................................ 203
LIMP
Default .................................. 189
Measuring capacitor ............................ 200
Measure coil ..................................... 200
Loopback ................................................. ... 16
Loopback cable .................................... 32
B
Baffle Step
Correction .............................................. 116
Bass Reflex
Level matching Vent ........................... 121
Volume flow method ......................... 124
M
Membrane
linear excursion .............................. 205
Xmax ................................................. .. 205
D
MessBox ................................................. .... 19
Measure
Data Export
Analytical measurement space .................................
Specify CSV format ......................... 102
94
Dodecahedron ................................................ 97
electrically .............................................. 138
DUT ................................................. .......... 13
Free field measurement ..................................... 82
Crossover ................................... 138
F
Ground Plane ......................................... 84
Crossover
Half-space ............................................... 85
Delay ................................................. ... 113
Measuring distance ......................................... 107
Load and Sum ...................................... 126
Measurement environment ......................................
80
H
Microphone comparison distortion ...................... 181
Anechoic chamber .......................... 83
Hall radius ............................................. 90, 95
Sound ........................................... 149
Shortcuts .............................................. 175
I
Measurement window
Impedance ................................................. ... 17
Gate ................................................. ....... 86
Accuracy .......................................... 202
Measuring chain ................................................. .. 41
Measurement error ............................................
Measuring line
202
Avoid interference ............................. 68
Measuring cable ............................................ 202 Measurement microphone
Measurement ............................................... 202
............................................ 17
Overlay ................................................. 131
Data ................................................. ..... 46
Impulse response ............................................ 77
SPL ................................... 46
Installation
Compensation frequency response errors ....... 53
Program ................................................. 7
Level calibration ................................... 45
Measuring room
K
Estimation of the measurement window .............. 93
Calibration ................................................. . 18
Measuring
single-channel ............................................... 24
Excitation signals .................................. 72
Loopback ............................................... 14
Transmit ................................................. .. 71
Measurement microphone ........................................ S / N Ratio .............................................. . 69
45
Amplifier ........................................... 61
Sound card ............................................. 42
zweikanlig .............................................. 21
N
THD
Reverberation time
Farina ................................................. .. 145
Estimation of the measurement period ..................
Sine ............................................. 145
98
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ARTA - Compendium
Setting the excitation time ........................ 98
Sound
automatic evaluation .................... 103
Rating filter .................................. 150
graphical representation ........................... 103
Simulation
Measurement setup ............................................ 97 AJ-Horn ............................................... 135
BoxSim ................................................ 105
Near field
Baffle Step ........................................... 116
Edge ................................................. .... 136
Level correction ........................................ 92
SEO ................................................. ..... 113
Positioning of the microphone ................ 91
Angle error ........................................ 106
Sound card ................................................. .. 9
tested cards ....................................... 9
O
Quality ................................................. . 36
Overlay
Noise level ........................................... 40
Frequency Response ....................................... 166 test ................................................. ..... 32
Impedance .............................................. 131
Sound Card Setup
Impulse response ...................................... 168
ASIO ................................................. ..... 31
Average Power ..................................... 173
WDM - Vista / Windows 7 ................... 29
Subtract ................................................ 173
WDM - Windows XP ............................ 27
Sound mixer
Setting .......................................... 15, 33
P
STEPS
Level
single-channel vs. dual-channel .................... 180
2 Pi ................................................ ....... 174
Default .................................. 176
4 Pi ................................................ ....... 174
THD vs. Amplitude ............................... 185
Far field ................................................ 174
Leq ................................................. ...... 149
Near field ................................................. 174
T
Q
Quasi-free field
Closed Box ........................................... 114
Vented Box .......................................... 120
R
RE
measure with multimeter ........................ 199
Resonance
Downsampling ..................................... 156
Housing ................................................ 158
Material ................................................ 159
Membrane .............................................. 161
Space ................................................. ... 154
S
Target
Filter functions ................................... 129
Menu ................................................. ... 128
Phase ................................................. ... 130
TSP
Select excitation level ....................... 197
measured in the test case ........................ 191
Measure ................................................. 197
Measurement location of the speaker .................
197
with additional mass measure ...................... 195
V
Connecting cable
Pin Assignment ..................................... 12
Comparison
Farina - STEPS .................................... 148
Save Text ................................................ .. 165
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