Polar Pattern Measurements - Medición de Patrones Polares

7/28/2019 Polar Pattern Measurements - Medicin de Patrones Polares

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Acoustics Instruments and Measurements April 2013, Caseros, Buenos Aires Province, Argentina

POLAR PATTERN MEASUREMENTS FOR A MECHANICAL ANDBIOMECHANICAL SOUND SOURCES

AGUSTN Y. ARIAS 1

1Universidad Nacional de Tres de Febrero, Buenos Aires, Argentina.

[email protected]

Abstract This work seeks to analyze and describe the radiation directional behavior of two sound sources with

completely different nature characteristics: a recording studio loudspeaker and a human voice (biomechanical

source). The procedures, considerations and limitations of the measurement process are explained as well as theobtained results are analyzed.

1. INTRODUCTIONThe polar pattern is one of the most important

characteristics of a sound source, allowing to known

the spatial behavior that the sounds will have in terms

of relative sound pressure levels and frequency. Such

behavior does not consider any acoustic phenomenon

linked to the room where the source is used, it refers

to the dynamics of irradiation in a so-called "freefield" where no sound reflections exists. In practice,

in order to obtain these patterns the measurements of

the sound source are performed in an anechoic

chamber, which are specially equipped to eliminate

any kind of reflection and to achieve completeisolation from external noise. This condition of"room without reflections and without noise" could

not be fulfilled in this report because the

measurements were made in a room with no acoustic

treatment used as classroom. Therefore we studied

various methodologies to minimize systematic errors

that would occur in the measurements, which will be

detailed later. It is important to remark that there is

no a standard which specifies the correct way to

perform polar patterns measurements. For these

reasons the results obtained do not have a high degree

of accuracy, although can be used as very good

approximations to the actual behavior of the sound

sources. For this work a recording studio loudspeaker

and human voice were used.

2. ROOM CHARACTERISTICSThe dimensions of the classroom where the

measurements where performed are 9.31 x 9.34 x

3.21 meters, with a double-layer ceiling with 60 cm

air cavity between as seen in Figure 1. There were a

total of fifty chairs placed on the periphery of the

room and a table on where it was installed thecomputer used for audio recording. The side wall

exposed to the outside of the building has fiveWindows that remained closed during the tests.

Figure 1: Classroom where measurements were made.Loudspeaker and biomechanical sound source position.

Circular array for the microphones.

3. EQUIPMENT USED FOR THEACQUISITION OF SOUND RECORDINGS

For both sources measurements, it was used the

same data acquisition system. The following list

details the equipment and software employed:

Notebook

M-Audio Fastrack audio interface

XLR-XLR cables

Microphone tripods

Svantek model Svan959 sonometer

Extension cable for the sonometers

microphone

Adobe Audition

o Sample rate: 44100 Hz

o Resolution: 24 bits

o Channel mode: Mono

AURORA plugins

EASERA


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In addition, only for the biomechanical

measurements it was used a turntable OUTLINEmodel ET250-3D.

4. LOUDSPEAKER MEASUREMENTSIn the first instance, measurements for the

loudspeaker were made.

4.1Loudspeaker CharacteristicsThe loudspeaker measured is a KRK

SYSTEMS Rokit8 model as shown in Figure 2.

Figure 2: Rokit8 loudspeaker used for the polar pattern

measurement.

Its most important characteristics provided by the

manufacturer are:

Configuration: 2-Way

System type: Active Studio Monitor

Low-Frequency: 8" Aramid Glass Compositewoofer

Mid-Frequency: N/A

High-Frequency: 1" soft dome tweeter

Frequency Response: 44Hz - 20kHz

Max Peak SPL: 109 dB

Amplifier Class: Class A-B

Power Output: 90W

High Frequency: 20W

Mid-Frequency: N/A

Low Frequency: 70W

Input Impedance (Ohms): 10 K Ohm

balancedHF Level Adjust:-2dB, -1dB, 0, +1dB

System Volume: (-30dB - +6dB)

Enclosure Construction: MDF

Dimensions (H x W x D): 15.5" (394mm) x

10.83" (275mm) x 11.73" (298mm)

Weight: 26 Lbs. (12 Kg.)

This loudspeaker consists of one woofer and atwitter to cover the complete audible spectrum. It is

mainly used for near field monitoring in recording

studios.

4.2 Microphones characteristicsThe two microphones used for the recordings

were EARTHWORKS model M50. Their most

important characteristics provided by the

manufacturer are:

Frequency Response: 3Hz to 50kHz 1/-3dB

Polar Pattern: Omnidirectional

Sensitivity: 30mV/Pa (-30.5dBV/Pa)

Power Requirements: 48V Phantom, 10mA

Max Acoustic Input: 142dB SPL

Output: XLR (pin 2+)

Output Impedance: 100, balanced (50 ea.

pin 2 & 3)

Min Output Load: 600 between pins 2 & 3

Noise: 22dB SPL equivalent (A weighted)

Dimensions L x D: 229mm x 22mm (9 x.860 inches)

Weight: .5 lb. (225g)

Figure 3: Earthworks M50

4.3 Loudspeaker and microphones locationThe loudspeaker was placed at a distance of 4.71

meters from the lateral walls and 3.71 meters from

the front wall (which supports the board), as shown in

Figure 4, to minimize the coloration effects at lowfrequencies due to the "eigenmodes" of the room [1]

and to maximize de arrival time of the first reflection.The loudspeaker was oriented with its front face

pointing towards the center of the room.

Figure 4: Loudspeaker Location

The loudspeaker is held fixed during the entire

measurement process while the microphones varying

its position around it. To accomplish this, the floor

was marked with different microphone positions

following a circular array centered at the geometricalcenter of the base of the loudspeaker and a radius of


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90 cm. That distance was used to place the

microphones, ensuring that it works in the activezone of the loudspeaker [2]. The measurement points

were defined by steps of 10 degrees, the degree zero

corresponding to the longitudinal and symmetrical

location of the front face of the loudspeaker. The

microphones (and the loudspeaker too) were placedon the floor to minimize the effects of cancellationgenerated by the floor reflections, known as "comb

filter" [3].

4.4 Measurement signalTo measure the polar pattern of the loudspeaker it

was decided to use a "Long-sine sweep", defined in

equation (1).

This type of signal has some advantages that

favor the measure, primarily considering the signal-

to-noise ratio is achieved. Because this type of signal

increases in frequency proportionally with time, all

the sound energy is concentrated moment to moment

in a single frequency, producing a markedimprovement in the signal-to-noise ratio compared to

other types of signals that distribute all their energy

in the entire spectrum at each instant of time (white

noise, pink noise, MLS) [4]. The AURORA plugin

allows generating the sine sweep and simultaneously

creates the inverse filter of the same that is used with

the signal measured by the microphone to perform

the convolution between them and thus obtain the

impulse response of the loudspeaker [4]. The decision

of using this method is that once the impulse

responses is obtained at each measurement point, we

can apply a temporal windowing to remove all

spectral information generated by the room

(reverberation) as it is explained below. It was

generated, then, a sine sweep signal with the

following characteristics:

Bandwith: 8012000 [Hz]

Duration: 10 [s]

Time increase type: Exponential

Fade-in / Fade-out: 0,1 [s]

Silence: 2 [s]

The exponential time increase gives more

duration to the low frequency time section regarding

the high frequencies sweep duration, allowing auniform energy distribution in time and frequency.

4.5 Sine sweep recordings for both Horizontaland Vertical Planes.

Using the microphones distribution abovementioned, it was proceeded to the sine sweep

measurements at each point. Twenty measurements

were recorded between 0 and 180 for the horizontalplane with a spacing of 10, since the loudspeaker is

considered as a horizontally symmetric radiation

source. One microphone measured between 0 and

90 while the other did the same between 100 and

180. In 180 there was an additional recording withthe other microphone to evaluate the difference

between them and adjusting the levels of the recorded

audio signal.

For the vertical plane the procedure was similar,

but with the loudspeaker supported on one of its

lateral faces (lying). In this position it cannot be

assumed a symmetrical distribution of the sound

field, thus it was measured the 360 full

circumference with 10 separation. As with the

horizontal plane, each microphone measured half the

entire circumference and one measuring was repeated

at 0 for the same compensation of levels as in the

previous case.

Figure 4: Loudspeaker and microphones positions for

the horizontal plane measurements

Figure 5: Loudspeaker and microphones positions for

the vertical plane measurements

4.6 Background noise evaluation.Although we already know the advantages

provided by the sine sweep with respect to signal to

noise ratio, an evaluation of the background noise (it

will be useful for the human voice measurements). It

was measured for 3 minutes with the sound level

meterin mode equivalent sound pressure level and

obtained a Leq value of 46 dBA. To improve the


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results, it was chosen the maximum loudspeaker

volume (defined as "+6 dB" by the manufacturer) butit was not modify the HF gain, letting it in 0dB

position.

5. HUMAN VOICE MEASUREMENTSThe second part of this study was to measure the

polar pattern of a male singer.

5.1Characteristics of the recorded singing.The singer was requested to sing in three different

scales a C note, the first corresponding to C 130.81

Hz, the second to C 261.63 Hz and the last to C

523.25 Hz.

5.2 Testing MethodologyThe procedures for measurements were different

to those made for the loudspeaker. Firstly, it was used

a measurement microphone in a fixed position (0) ata distance of 60 cm from the mouth of the singer and

a height of 1.30 meters corresponding to the height of

the mouth, and placed the singer on a turntable

(Outline ET250-3D). On the same platform it was

placed another microphone so as to rotate with the

singer, always maintaining a constant distance with

his mouth (Figure 6). This reference microphone is

used to adjust the levels with the fixed-position

microphone, as it is explained below.

At the same time, the sound level meter

microphone was placed at a distance of 2 mm from

the fixed-position microphone, using a 10 meter

extension cable (Figure 7). The purpose of this is to

evaluate the signal to noise ratio between the singer's

voice and the background noise, which affects

directly the effectiveness of the results (besides thecontribution of the reverberant room). If for some

reason the level difference between the singer's voice

and the background noise was less than 8 dB, the

measurement was repeated.

Figure 6: Singer, microphones and turneable positions.

Figure 7: Location of the sound level meter microphone.

Once all the necessary elements were located it

was proceeded to take the measurements. The singer

and the reference microphone were rotated in steps of

10 from the position 0 to 350. At each step the

three scales of the sung note C were recorded by bothmicrophones and the background noise was evaluated

during the recording session.

6. ANALYSIS AND CONSIDERATIONSOnce acquired all sound recordings from both

sources, corresponding analyzes were performed

considering the factors that impacted negatively on

the measurement processes.

6.1 Analysis of the loudspeakerThe recordings obtained at every point between

0 and 180 for the horizontal plane and between 0and 350 for the vertical plane of the sine sweep

signal were processed as follows:

First, it was necessary to adjust the audio

recordings from both microphones in order to

eliminate the sensibility differences between them

and equate the signals amplitudes. As it is described

in 4.5, both microphones performed a measurementin the 0 position for the vertical plane. To equate theamplitude levels in that position, it was decided to

seek the maximum amplitude measured by each

microphone and then amplify the lowest until it

matches to the higher. This allows us to know which

microphone produce lower signal levels and howmuch lower is regarding the other microphone. Thisamplification process was performed for the nine

positions in which that microphone was used. The

same procedure was performed for the horizontal

plane (in this case, the 180 position was measured

with both microphones). Then, the impulse response

of the loudspeaker at each position was obtained byconvolving the recorded signals with the inverse sine

sweep filter using the AURORA plugins (see Anex).

These impulse responses should be enveloped by

applying a temporal windowing to remove all

information relating to the reverberant field of the

room. Thus it was decided to silence the signal from


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the arrival of the first reflection. This process was

done manually by observing the waveforms of eachimpulse response as shown in Figure 8.

Thus it was possible to obtain the impulse

response of the loudspeaker minimizing sound

contribution due to room reflections. All impulse

responses were loaded and analyzed spectrally withEASERA software. It was used the "ExportSpectrum" option to load the data into an Excel

spreadsheet so as to generate the polar plots with the

spectral values filtered in octave bands between 125

and 8000 Hz. The results below 125 Hz could not be

taken into account because of background noise was

comparable to the maximum level of amplification of

the loudspeaker for this frequency range. This is the

reason why the lower frequency of the sine sweep is

80 Hz. Figure 9 shows the spectrum of an impulse

response analyzed with EASERA.

Figure 8: Impulse response for one position in theloudspeaker analysis. Top: Original impulse response.

Bottom: Time windowing applied to the impulse response.

Figure 9: Spectrum of an impulse response in EASERA

It is necessary to make an important consideration

on these measurements about the acoustic center ofthe source. To perform a highly accurate polar pattern

it is required a microphone location such that it point

towards to the center of the acoustic source. It was

chosen as the center for the measurement the

geometrical center of the loudspeaker, as there wasno information about the acoustic center, so the polardiagram curves must be adjusted manually to avoid

asymmetry problems. The acoustic center of a

loudspeaker varies depending on the frequency and

also it must be considered that this loudspeaker

consists of a woofer and a twitter (different acoustics

centers)[5]. This consideration produces systematic

errors in the measurement process.

6.2Analysis of the human voiceThe recordings obtained at every point between

0 and 350 of the three C scales were processed as

follows:Firstly, each recording was divided into three

parts, separating into individual audio files each

recording of the C note according to their key (Figure

10). In this manner the levels differences can be

adjusted individually. The adjustments consisted of

find the maximum value of each recording made by

the reference microphone (the one that rotated with

the singer) in each position between 0 and 350. The

amplitude of the signal in the 0 position was taken as

the reference value. All others signals (between 10

and 350) were leveled to that reference value,

attenuating or amplifying. And then, the same

amplitude adjustments were applied to the

measurement microphone (the one that was kept in

a fixed position) for each position. It should be noted

that not always the maximum value of amplitudebetween different measurement positions have the

same instant of time, suggesting that the analysis

leveling errors will affect the final results. A longer

study is required to analyze temporal variation of

maximum values and seek for better solutions.

Figure 10: Signal divisions

To analyze each signal individually, it was

proceeded analogously to the loudspeaker analysis by

using the "Export Spectrum" function of EASERA.

Figures 11 to 13 show three spectra corresponding to

First reflection without

temporal windowing

First reflection after

temporal windowing

C-130.81 Hz C-261.63 Hz C-523.25 Hz


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a C at each of its tuning to the 0 position. It can be

observed the presence of the fundamental tone, itsharmonics and the background noise contribution.

Figure 11: Spectrum of C 130.81 Hz



The most important factor that is observed in

these graphs is the background noise level which is

comparable with the source levels. This situation is

further exacerbated for those recordings made with

the subject positioned back to the measurement

microphone, as the C of 130.81 Hz cannot exceed by

more than 10 dB the background noise according to

the sound level meter measurements, which

introduces serious errors in the final results. Another

problem that introduces this method of measurement

is the degree of precision regarding the singer's pitch,which can be seen in each position as tuning slight

variations for the three notes. It cannot be possible to

eliminate de reverberant contribution of the room

with this method neither. Knowing these limitations,

the spectral analysis of source directivity was limited

and only took into account the values given in the

bands of 125 Hz (close to C 130.81 Hz), 250 Hz

(close to C 261.63 Hz) and 500 Hz (near C 523.25

Hz). Increase spectral range of analysis introduce

levels errors which, added to the limitations

mentioned above, would deliver in values that do not

correspond with reality.

7. RESULTSThe results of the analysis are presented for both

sound sources:

7.1Loudspeaker polar pattern results:Figures 14 and 15 shows the graph obtained for

the polar patterns in the horizontal plane.

Figure 14: Polar patter of the loudspeaker. Horizontalplane. 125, 250, 500 and 1000 Hz octave bands.

Figure 15: Polar patter of the loudspeaker. Horizontal

plane. 2000, 4000 and 8000 Hz octave bands.

From these results several conclusions may be

drawn. Directivity behavior according to the working

frequency shows how the loudspeaker directivity

increases with frequency, tending to be more

omnidirectional for low frequencies. Whereas the

acoustic center of the source at lower frequencies is afew inches past the physical center, sound pressure

levels remain fairly constant in all directions in the

bands of 125 and 250 Hz. It also shows a noticeable

drop of about 3 dB at 8000 Hz on the front radiation

axis (0). This level can be adjusted manually

between -2, -1, 0 and +1 dB offset so as to accentuate

Background noise C 130.81 Hz




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or attenuate the SPL. During the measurements, this

value was maintained at 0 dB.

Finally, considering the above mentioned, it

denotes a homogeneous power distribution for allbands in front of the loudspeaker addresses (between

90, 0 and 270). This ensures a good listingsituation regarding the energy balance. Figures 16

and 17 shows the graph obtained for the polar

patterns in the vertical plane.

Figure 16: Polar patter of the loudspeaker. Vertical lane.

125, 250, 500 and 1000 Hz octave bands.

Figure 17: Polar patter of the loudspeaker. Verticalplane. 2000, 4000 and 8000 Hz octave bands.

As we can see, the directivity behavior in the

bands of 125 and 250 Hz is similar to that obtained in

the horizontal plane, and meets the trend to be

omnidirectional considering moving the acoustic

center. As frequency increases, the sound radiation ofthe source becomes more directive on the front axis

(0) with a significantly attenuation of the radiation

level from the rear face of the source (between 15 to

25 dB for 4000 Hz) due to the loudspeakers

enclosure.It is important to note what happens between

positions 330 and 350 in the band of 2000 Hz,

where a marked fall in the sound pressure level is

appreciated (about of -5dB). It is estimated that this

band is the junction between the crossover filters andbecause the microphone position measurement acancellation occurs between the phases of the

reproduced signal. There was no official information

on this matter, so that there should be further

investigation to verify this hypothesis.

7.2Biomechanical source polar pattern results:Figures 18 to 20 show the polar pattern obtained

for the horizontal plane of the human voice recorded.

For the band of 125 Hz, it can be seen that the

maximum sound pressure levels were measured in

the lateral directions, not in the front direction as one

might expect. However, the differences are verysmall (less than 3 dB) and can be considered that the

radiation is omnidirectional with the exception of the

positions 160 and 200, wherein we can see a

marked attenuation of the sound field of about 6 dB.

In the 500 Hz band we can appreciate the attenuation

of sound pressure level at the rear due to the singer's

acoustic shadow generated by the head, while the

frontal lobe takes a more directive shape comparing

to the band of 125 Hz

The measurement difficulties mentioned in 6.2that could not be offset or eliminate produce errors on

these results as can be seen from polar patterns of

speech in 250 and 500 Hz. It is observed that the

sound pressure level in the back side of the singer's

mouth is much lower (about 10 dB) at 250 Hz

compared to 500 Hz, when in reality this is not so [6].The reasons that cause this type of error may be due

to reflections within the enclosure, which generate

constructive and destructive interference. In addition,

displacements out of the referral center, that occurs

when the singer flips around the circle, makes

difficult the correct location of the polar curveswithin the graph axes.

Figure 18: Polar patter of the shuman voice. Horizontalplane. 125 Hz octave band.


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8. CONCLUSIONSThe measurement and analysis methodologies

employed and the results obtained deserve to be

studied. On one hand, it was found that themethodology used to measure the polar pattern of the

loudspeaker generates reliable results because, as it

was described, applying a temporal window to

eliminate the reflections in the impulse responses

allow us to become independent of the room

influence. In addition, the use of the Log-Sine sweep

signal to obtain the impulse responses improves thesignal-to-noise ratio, minimizing the background

influence too. It achieves appreciate clearly the

omnidirectional trend of the source for low frequency

(up to 500Hz) and, conversely, a directional radiation

tendency for higher frequencies (above 500Hz). The

physical foundations associated with this "directional

behavior" have been widely studied and

demonstrated by L. Beranek [7]. The difference of

levels in each frequency band can be due to various

factors, such as air absorption at high frequencies, theloudspeaker response itself, distance between source

and microphone, etc. However, the method can be

used to get a rough idea of the actual behavior of the

sound source.In the other hand, the results obtained for the

polar pattern of the biomechanical source are more

affected by external conditions imposed by the

environment in which the measurements were carried

out. Variations in pitch of the singer, reflectionsinterference and displacements of the acoustic centeralso generated systematic errors in the measurements.

Another important consideration is referred to the

signals analyses. As it was explained, the maximum

value for each C tone was taken as the reference for

the level adjustment, but those values not always

correspond to the same time position for the same C

tone in different measurements positions, so there is a

deviation according to the actual polar pattern values.

The last consideration is about the background

noise. The noise levels detracted the dynamics of the

recordings and contaminated the spectral analysis of

the voice. At the moment of the human voicerecordings, there was a lengthy impact noise due to

the work of some workers who handled hammers on

a wall of a construction site, besides the train noise.

The background noise mainly affects the low

frequency results (125Hz). The results obtained mustnot be taken in account because they are too much

influenced by the external conditions (rooms

reflections, background noise) and by the signal post-

processing. For all these reasons, it is required better

external conditions to perform this type of

measurements for a human voice. An anechoicchamber is the best option due to its absence of

reflective field and noise isolated characteristics. Inaddition a set of several microphones is also

recommended to avoid the leveling process.

9. REFERENCES[1] Ballou, Glen. Modal Room Resonances.

Handbook for Sound Engineers. Cap. 6, pp. 128-137.

Focal Press.

[2] Keele, Jr., D. B. 1974. Low-Frequency

Loudspeaker Assessment by Nearfield Sound-Pressure Measurement. JAES Volume 22 Issue 3 pp.

154-162.

[3] Wikipedia, Online enciclopedia. Comb filter.

http://en.wikipedia.org/wiki/Comb_filter.

[4] Farina, Angelo. Impulse Response

Measurements by Exponential Sine Sweeps.Parma,

18 October 2008.

[5] Vanderkooy, John. The Acoustic Center: A New

Concept for Loudspeakers at Low Frequencies.

AES:121 (Oct 2006) Paper:6912.

[6] Marshal, A. H., Meyer, J. The Directivity andAuditory Impressions of Singers.

[7] Beranek, Leo. Acoustics. American Institute of

Physics; Editin: Rev Sub (1986).


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ANEXAURORA PLUGINS

Aurora is a suite of plug-ins for Adobe Audition: acoustic impulse responses of rooms can be measured and

manipulated, to recreations dimensional simulations of acoustic and acoustic space.

Powered by Angelo Farina, its use allows rapid and effective assessment of the impulse responses acquired in the

room. Such responses are obtained by recording signals generated by the same plug-in.

Aurora plugins are XFM modules, created for being use as "native" plugins of Adobe Audition (formerly

CoolEditPro).

The XFM format was created by Syntrillium, the developer of CoolEdit, as their "native" plugin format.

Currently, if you explore the directory where Audition is installed, you will see dozens of files with the XFM

extension, which perform most of the "effects" already coming with the standard Audition package.

From a technical point of view, an XFM plugin is simply a DLL (Dynamic Linked Library), with the extension

changed from DLL to XFM. In practice, for developing these XFM plugins, a specific API (Application

Programming Interface) must be employed. The CoolEdit/Audition API was originally available form the

Syntrillium web site, but later Adobe dropped its availability, and so it is now not anymore officially possible to

create new XFM modules.And this is a pity, because the other formats for external plugins, such as Active-X (Microsoft) and VST (Steinberg),

although supported by Adobe Audition, do not allow to perform many of the tricks made possible employing XFM

plugins.

This is the main reason for which, albeit the XFM format is not anymore supported officially from Adobe, theAurora plugins are still in XFM format, instead of having being recompiled as VST plugins.

It is still possible that, in the future, the Aurora plugins are ported in other formats. Some modules have been

already ported under Audacity, for example, and are currently under Beta test.

Keep a look on this forum for the latest new about porting Aurora to other formats...

A. Farina

Long-Sine Sweep generation

The function x(t) is a band-limited signal of the sine sweep, whose frequency varies exponentially with time from f1

to f2. It is defined as follows:


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Application:

Selecting Generate/Sine sweep in the Adobe Audition Menu, the sine sweep is generated once the frequency

range and duration settings were made, and it is reproduced through the loudspeaker:

Once this signal is created, the same program stored in the "windows clipboard" the inverse filter z(t), which is usedto perform the deconvolution process to obtain the impulse response.

Inverse filter z(t)

Then it proceeds to measure the signal reproduced by the loudspeakers. We call this signal y(t):


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Recorded Signal

The nonlinear behavior of the loudspeaker produces the appearance of harmonics.

Then, selecting Aurora/Convolve with clipboard function in the Effects menu, the deconvolution for theimpulse response is obtained by convolving the signals z(t) and y(t):

IR = z(t) * y(t)

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Polar Pattern Measurements - Medición de Patrones Polares

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