Acoustics Instruments and Measurements

April 2013, Caseros, Buenos Aires Province, Argentina

POLAR PATTERN MEASUREMENTS FOR A MECHANICAL AND BIOMECHANICAL SOUND SOURCES

AGUSTN Y. ARIAS 1
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Universidad Nacional de Tres de Febrero, Buenos Aires, Argentina. agustin.arias@outlook.com

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 the obtained results are analyzed.

1. INTRODUCTION The 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 "free field" 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 complete isolation 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 CHARACTERISTICS The 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 the computer used for audio recording. The side wall exposed to the outside of the building has five Windows 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 THE

ACQUISITION 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

In addition, only for the biomechanical measurements it was used a turntable OUTLINE model ET250-3D. 4. LOUDSPEAKER MEASUREMENTS In the first instance, measurements for the loudspeaker were made. 4.1 Loudspeaker Characteristics The loudspeaker measured is a KRK SYSTEMS Rokit8 model as shown in Figure 2.

4.2 Microphones characteristics The 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 2: Rokit8 loudspeaker used for the polar pattern measurement. Figure 3: Earthworks M50

Its most important characteristics provided by the manufacturer are: Configuration: 2-Way System type: Active Studio Monitor Low-Frequency: 8" Aramid Glass Composite woofer 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 balanced HF 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 a twitter to cover the complete audible spectrum. It is mainly used for near field monitoring in recording studios.

4.3 Loudspeaker and microphones location The 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 low frequencies 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 geometrical center 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 active zone 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 placed on the floor to minimize the effects of cancellation generated by the floor reflections, known as "comb filter" [3]. 4.4 Measurement signal To 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 signalto-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 marked improvement 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: 80 12000 [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 a uniform energy distribution in time and frequency. 4.5 Sine sweep recordings for both Horizontal and Vertical Planes. Using the microphones distribution above mentioned, it was proceeded to the sine sweep

measurements at each point. Twenty measurements were recorded between 0 and 180 for the horizontal plane 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 with the 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 meter in 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) but it was not modify the HF gain, letting it in 0dB position. 5. HUMAN VOICE MEASUREMENTS The second part of this study was to measure the polar pattern of a male singer. 5.1 Characteristics 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 Methodology The procedures for measurements were different to those made for the loudspeaker. Firstly, it was used a measurement microphone in a fixed position (0) at a 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 the contribution 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 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 both microphones and the background noise was evaluated during the recording session. 6. ANALYSIS AND CONSIDERATIONS Once 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 loudspeaker The recordings obtained at every point between 0 and 180 for the horizontal plane and between 0 and 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 measurement in the 0 position for the vertical plane. To equate the amplitude 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 how much lower is regarding the other microphone. This amplification 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 by convolving 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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Figure 6: Singer, microphones and turneable positions.

the arrival of the first reflection. This process was done manually by observing the waveforms of each impulse 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 with EASERA software. It was used the "Export Spectrum" 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.

It is necessary to make an important consideration on these measurements about the acoustic center of the 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 was no information about the acoustic center, so the polar diagram 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. 6.2 Analysis of the human voice The 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 amplitude between 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.

First reflection without temporal windowing

First reflection after temporal windowing

Figure 8: Impulse response for one position in the loudspeaker analysis. Top: Original impulse response. Bottom: Time windowing applied to the impulse response.

C-130.81 Hz

C-261.63 Hz

C-523.25 Hz

Figure 10: Signal divisions

Figure 9: Spectrum of an impulse response in EASERA

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
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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, its harmonics and the background noise contribution.

mentioned above, would deliver in values that do not correspond with reality. 7. RESULTS The results of the analysis are presented for both sound sources: 7.1 Loudspeaker polar pattern results: Figures 14 and 15 shows the graph obtained for the polar patterns in the horizontal plane.

Background noise

C 130.81 Hz

Figure 11: Spectrum of C 130.81 Hz

Background noise

C 261.63 Hz Hz

Figure 12: Spectrum of C 261.63 Hz Figure 14: Polar patter of the loudspeaker. Horizontal plane. 125, 250, 500 and 1000 Hz octave bands.

Background noise

C 523.25 Hz Hz

Figure 13: Spectrum of C 523.25 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

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 a few 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
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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 all bands in front of the loudspeaker addresses (between 90, 0 and 270). This ensures a good listing situation regarding the energy balance. Figures 16 and 17 shows the graph obtained for the polar patterns in the vertical plane.

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 and because the microphone position measurement a cancellation 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. 7.2 Biomechanical 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 very small (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 6.2 that 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 curves within the graph axes.

Figure 16: Polar patter of the loudspeaker. Vertical lane. 125, 250, 500 and 1000 Hz octave bands.

Figure 17: Polar patter of the loudspeaker. Vertical plane. 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 of the 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

Figure 18: Polar patter of the shuman voice. Horizontal plane. 125 Hz octave band.
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Figure 18: Polar patter of the shuman voice. Horizontal plane. 250 Hz octave band.

Figure 18: Polar patter of the shuman voice. Horizontal plane. 500 Hz octave band.

8. CONCLUSIONS The measurement and analysis methodologies employed and the results obtained deserve to be studied. On one hand, it was found that the methodology 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 the signal-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, the loudspeaker 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, reflections interference and displacements of the acoustic center also 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 voice recordings, 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 must not be taken in account because they are too much influenced by the external conditions (rooms reflections, background noise) and by the signal postprocessing. For all these reasons, it is required better external conditions to perform this type of measurements for a human voice. An anechoic chamber is the best option due to its absence of reflective field and noise isolated characteristics. In addition 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 SoundPressure 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 and Auditory Impressions of Singers. [7] Beranek, Leo. Acoustics. American Institute of Physics; Editin: Rev Sub (1986).

ANEX AURORA 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, the Aurora 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 used to 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 the impulse response is obtained by convolving the signals z(t) and y(t): IR = z(t) * y(t)

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