Equalization filter design for downfiring flat television speakers

Many loudspeakers in newer flat television systems are mounted on the bottom of the television, with their diaphragms facing downward, so as to be hidden inside the TV frame. This kind of loudspeaker installation, called downfiring, induces relatively large reflections from the walls of a room. The increased reflections inevitably change the impulse response of the loudspeakers, which leads to a perceptible distortion in sound quality. In this study, an equalization procedure to resolve the distortion due to downfiring loudspeakers is presented. The change in the early-arriving sound from the room reflections was analyzed, and the specific experimental environment was set up to measure the early reflections for designing a single-channel equalization filter. The inverse technique for the single-input multiple-output (SIMO) system was applied such that both the magnitude and phase responses from downfiring loudspeakers can be equalized at multiple listener positions. We also investigated the possible equalization range in space and time, for which the single-channel filter can work effectively. The performance of the filter designed for the equalization range was then demonstrated by experiments. The experiments were performed using a flat television mock-up installed in a reflective environment. For the performance evaluation of the proposed method, we employed two different measures that can represent the magnitude and phase distortions: the mean squared error of the magnitude and an energy decay curve. The experimental results demonstrate that the single-channel filter can reduce the early reflections from a downfiring loudspeaker by a considerable amount.     

Study of vertical sound image control with parametric loudspeakers

A parametric loudspeaker utilizes nonlinearity of a medium and is known as a super-directive loudspeaker. In this paper, the sound localization in the vertical direction using the upper and lower parametric loudspeakers is confirmed by listening tests and physical measurements. The differences in levels between the upper and lower parametric loudspeakers are varied as a parameter. The direction of sound localization in the vertical plane can be controlled not only when the acoustical axis is set to the right ear but also when it is set to at 5 deg to the right of the right ear. The effect of the level difference between the upper and lower loudspeakers is weaker than the differences observed when using ordinary loudspeakers. We obtained interesting characteristics of the left-right sound localization in the horizontal plane with the upper and lower parametric loudspeakers in the vertical plane. It is found that by setting the parametric loudspeaker at the right ear (that is, the horizontal angle of a listener to it is only 3 deg to the right), the direction of sound localization in the horizontal plane moved approximately 10 deg to the right. Moreover, by setting the parametric loudspeaker 5 deg to the right, the direction of sound localization moves approximately 20 deg to the right. The ILD (Interaural Level Difference) using a dummy head is calculated from the measured left and right sound signals. It is determined that ILDs of the parametric loudspeaker are larger than those of the ordinary loudspeaker. A simple geometrical acoustic model is introduced and analyzed. The analysis helps to explain the measured characteristics.     

Sound field control with a circular double-layer array of loudspeakers

This paper describes a method of generating a controlled sound field for listeners inside a circular array of loudspeakers without disturbing people outside the array appreciably. To achieve this objective, a double-layer array of loudspeakers is used. Several solution methods are suggested, and their performance is examined using computer simulations. Two performance indices are used in this work, (a) the level difference between the average sound energy density in the listening zone and that in the quiet zone (sometimes called "the acoustic contrast"), and (b) a normalized measure of the deviations between the desired and the generated sound field in the listening zone. It is concluded that the best compromise is obtained with a method that combines pure contrast maximization with a pressure matching technique.     

Efficient equalization of multi-exciter distributed mode loudspeakers

An efficient digital equalization method is applied successfully to the problem of spectral equalization of multi-exciter distributed mode loudspeakers (DML). It is based on a chain of second-order sections of infinite impulse response parametric filters with very low computational cost. The method compensates for the measured multi-exciter DML response in order to achieve a desired frequency response. The sound radiation of these flat loudspeakers is a complex superposition of excited modes that vary strongly with frequency. Therefore, the characteristic multi-exciter DML spectrum is very irregular and is equalized with the method presented here for a natural, uncolored response. In multichannel systems, such as wave field synthesis (WFS), the use of efficient filters to equalize a large amount of drivers is an advantageous approach. The equalization process has been applied to two multi-exciter DML prototypes, comprising three and five exciters per panel. Both panel and exciter equalization have been addressed, which consequences on the filtered responses are discussed. Finally, some subjective assessments are carried out to optimize the order of the filter while maintaining the perceived quality of the equalization.     

Sound localization of stereo reproduction with parametric loudspeakers

A parametric loudspeaker radiates an audible signal by the interaction of the primary wave that is amplitude modulated and is known as a super-directivity loudspeaker. The parametric loudspeaker is one of the prominent applications of nonlinear acoustics. So far, the applications have been limited monaural reproduction sound system for public address in museum, station, street etc. In this paper, we investigated sound localization of stereo reproduction using two parametric loudspeakers in comparison with that using two ordinary dynamic loudspeakers. In subjective tests, the binaural information ILD (Interaural Level Difference) or ITD (Interaural Time Delay) was focused on. To investigate the characteristics of sound localization in a wide listening area, three typical listening positions were picked up. Signals were 500 Hz, 1 kHz, 2 kHz and 4 kHz pure tones and pink noise. The used parametric loudspeaker was an equilateral hexagon. The subjective test led to the results that when the parametric loudspeakers were used, the listeners at the three typical listening positions perceived the correct sound localization of not only pure tone but also pink noise and when the ordinary dynamic loudspeakers were used, except for the case of pure tone with ITD, the tendency was almost similar to those using the parametric loudspeakers. The second subjective tests were conducted in order to investigate in details the difference between parametric loudspeakers and ordinary dynamic loudspeakers by increasing the number of subjects. In the case of ITD and 500 Hz using the ordinary dynamic loudspeakers, three types of sound localization were categorized, in which the reversed type was major and the normal and the other types were minor. The ILDs which were measured with a dummy head and were calculated with several formulas were almost the same and indicated the reasons of the reversed typed sound localization and a serious influence of the crosstalk. It was found that in the case of pure tone with ITD, the contradiction between the binaural information ILD and ITD is remarkable, because the directivity of the ordinary dynamic loudspeakers was so dull that the crosstalk components had a serious influence on sound localization. It was determined the parametric loudspeaker could transmit correct binaural information to the listener, because the directivity of the parametric loudspeakers was so sharp that it suppressed the cross talk components.     

Thermally tunable EDFA gain equalizer using point symmetric cascaded Mach-Zehnder coupler

In this paper, a thermally tunable EDFA gain equalizer filter based on point symmetric cascaded Mach-Zehnder (CMZ) filter based two mode interference (TMI) coupler is presented with its mathematical model. Transmission characteristics of these CMZ couplers are analyzed and compared with Y symmetric CMZ couplers by using this model. For EDFA gain equalizer, point symmetric CMZ circuit is chosen due to its higher wavelength flattening width than Y symmetric CMZ circuit. The ripples of equalized EDFA gain spectrum are formulated and estimated from the equalized gain spectrum of point symmetric CMZ filters. It is found that 2 stage point symmetric CMZ coupler with binomial coupler distribution (2PB CMZ) using Δn = 5% provides gain equalized width of 35 nm with ripple of 0.4-0.6 dB and bending loss of 0.24 dB and device length is ∼15 times lower than that of the existing EDFA gain equalizer based CMZ filter. It is also seen that if during the fabrication process, waveguide core width w is increased or decreased by 0.1 μm (in percentage ∼±6.6%), the power imbalance of TMI based 2PB CMZ filter is slightly increased by ∼8% in comparison to that based on directional coupler (DC) by 40%. Low power thermooptic structure of varying gap between two waveguide cores with silicon trench just below the heater is used and it requires ∼1.5 times less heating power than the conventional structure for thermal tuning of EDFA gain equalization.     

Reverberation time measurements in non-diffuse acoustic field by the modal reverberation time

The increasing presence of low frequency sources and the lack of acoustic standard measurement procedures make the extension of reverberation time measurements to frequencies below 100 Hz necessary. In typical ordinary rooms with volumes between 30 m³ and 200 m³ the sound field is non-diffuse at such low frequencies, entailing inhomogeneities in space and frequency domains. Presence of standing waves is also the main cause of bad quality of listening in terms of clarity and rumble effects. Since standard measurements according to ISO 3382 fail to achieve accurate and precise values in third octave bands due to non-linear decays caused by room modes, a new approach based on reverberation time measurements of single resonant frequencies (the modal reverberation time) has been introduced. From background theory, due to the intrinsic relation between modal decays and half bandwidth of resonant frequencies, two measurement methods have been proposed together with proper measurement procedures: a direct method based on interrupted source signal method, and an indirect method based on half bandwidth measurements. With microphones placed at corners of rectangular rooms in order to detect all modes and maximize SNRs, different source signals were tested. Anti-resonant sine waves and sweep signal turned out to be the most suitable for direct and indirect measurement methods respectively. From spatial measurements in an empty rectangular test room, comparison between direct and indirect methods showed good and significant agreements. This is the first experimental validation of the relation between resonant half bandwidth and modal reverberation time. Furthermore, comparisons between means and standard deviations of modal reverberation times and standard reverberation times in third octave bands confirm the inadequacy of standard procedure to get accurate and precise values at low frequencies with respect to the modal approach. Modal reverberation time measurements applied to furnished ordinary rooms confirm previous results in the limit of modal sound field: for highly damped modes due to furniture or acoustic treatment, the indirect method is not applicable due to strong suppression of modes and the consequent deviation of the acoustic field from a non-diffuse condition to a damped modal condition, while standard reverberation times align with direct method values. In the future, further investigations will be necessary in different rooms to improve uncertainty evaluation.     

Adjustment on subjective annoyance of low frequency noise by adding additional sound

Structure-borne noise originating from a heat pump unit was selected to study the influence on subjective annoyance of low frequency noise (LFN) combined with additional sound. Paired comparison test was used for evaluating the subjective annoyance of LFN combined with different sound pressure levels (SPL) of pink noise, frequency-modulated pure tones (FM pure tones) and natural sounds. The results showed that, with pink noise of 250-1000 Hz combined with the original LFN, the subjective annoyance value (SAV) first dropped then rose with increasing SPL. When SPL of the pink noise was 15-25 dB, SAV was lower than that of the original LFN. With pink noise of frequency 250-20,000 Hz added to LFN, SAV increased linearly with increasing SPL. SAV and the psychoacoustic annoyance value (PAV) obtained by semi-theoretical formulas were well correlated. The determination coefficient (R²) was 0.966 and 0.881, respectively, when the frequency range of the pink noise was 250-1000 and 250-20,000 Hz. When FM pure tones with central frequencies of 500, 2000 and 8000 Hz, or natural sounds (including the sound of singing birds, flowing water, wind or ticking clock) were, respectively, added to the original sound, the SAV increased as the SPL of the added sound increased. However, when a FM pure tone of 15 dB with a central frequency of 2000 Hz and a modulation frequency of 10 Hz was added, the SAV was lower than that of the original LFN. With SPL and central frequency held invariable, the SAV declined primarily when modulation frequency increased. With SPL and modulation frequency held invariable, the SAV became lowest when the central frequency was 2000 Hz. This showed a preferable correlation between SAV and fluctuation extent of FM pure tones.     

Circular and rectangular via holes formed in SiC via using ArF based UV excimer laser

Circular via holes with diameters of 10, 25, 50 and 70 μm and rectangular via holes with dimensions of 10 μm × 100 μm, 20 μm × 100 μm and 30 μm × 100 μm and drilled depths between 105 and 110 μm were formed in 300 μm thick bulk 4H-SiC substrates by Ar/F₂ based UV laser drilling (λ = 193 nm) with a pulse width of ∼30 ns and a pulse frequency of 100 Hz. The drilling rate was linearly proportional to the fluence of the laser, however, the rate decreased for the larger via holes. The laser drilling produces much higher etch rates (229-870 μm/min) than conventional dry etching (0.2-1.3 μm/min) and the via entry can be tapered to facilitate subsequent metallization.     

Field measurement of airborne sound insulation between rooms with non-diffuse sound fields at low frequencies

Procedures for the field measurement of airborne sound insulation between rooms with diffuse fields are described in International Standard ISO 140-4. However, many dwellings contain rooms with volumes less than 50 m³, where low frequency measurements are less reliable; hence there is a need for a measurement procedure to improve the reliability of field measurements in rooms with non-diffuse fields. Procedures are proposed for sound pressure level and reverberation time measurements for the 50, 63 and 80 Hz third octave bands. The sound pressure level measurement combines corner microphone positions with positions in the central region of each room. This provides a good estimate of the room average sound pressure level with significantly improved repeatability.     

Improving speech intelligibility in classrooms through the mirror image model

Optimal classroom acoustical design can directly enhance students’ learning efficiency. Effective acoustical designs are important and necessary to achieve a high degree of speech intelligibility for listeners. A speech intelligibility metric, U₅₀, at different receiver positions in a classroom of 10 m × 8 m × 6 m was obtained by numerical simulations based on the mirror image model, with and without the uniform surface absorption coefficient. Comparisons show that increasing the absorption coefficient at the back wall can increase the speech intelligibility metric U₅₀ to the largest extent in the classroom. A numerical case study was then conducted in a typical classroom of 10 m × 10 m × 3.5 m, and the speech intelligibility was assessed through a third-order polynomial of Wonyoung and Murray [Wonyoung Y, Murray H. Auralization study of optimal reverberation times for speech intelligibility for normal and hearing-impaired listeners in classrooms with diffuse sound field. J Acoust Soc Am 2006;120(2):801-7].     

Directional perception of distributed sound sources

The perception of spatially distributed sound sources was investigated by conducting two listening experiments in anechoic conditions with 13 loudspeakers evenly distributed in the frontal horizontal plane emitting incoherent noise signals. In the first experiment, widely distributed sound sources with gaps in their distribution emitted pink noise. The results indicated that the exact loudspeaker distribution could not be perceived accurately and that the width of the distribution was perceived to be narrower than it was in reality. Up to three spatially distributed loudspeakers that were simultaneously emitting sound could be individually perceived. In addition, the number of loudspeakers that were indicated as emitting sound was smaller than the actual number. In the second experiment, a reference with 13 loudspeakers and test cases with fewer loudspeakers were presented and their perceived spatial difference was rated. The effect of the noise bandwidth was of particular interest. Noise with different bandwidths centered around 500 and 4000 Hz was used. The results indicated that when the number of loudspeakers was increased from four to seven, the perceived auditory event was very similar to that perceived with 13 loudspeakers at all bandwidths. The perceived differences were larger in wideband noise than in narrow-band noise.     

Sound source imaging of low-flying airborne targets with an acoustic camera array

Two-dimensional images of sound source distribution from near-ground airborne sounds are created using an array of 32 microphones and time-domain beamforming. The signal processing is described and array configurations spanning a square area with a side length of 3.45 m, approximately five wavelengths for a 500 Hz sound, are examined. Simulations of a 32-element under-populated log₆ × log₆ spaced array are given for sound sources centered over the array at 250 Hz, 500 Hz, and 1000 Hz. Stochastically optimized array geometry with a simulated annealing algorithm is discussed and a 32-element array optimized for a 500 Hz source is given along with a simulated image for direct comparison with the log₆ spaced array. Images from field testing a 32-element under-populated log₆ × log₆ spaced array are provided for a small aircraft flyover. Results show that this type of acoustic camera generates accurate images of sound source location. Suggested uses include monitoring small aircraft flying too low to be detected by radar as well as monitoring ecological events, such as bird migration.     

Apparent mass of the human body in the vertical direction: Inter-subject variability

The biodynamic responses of the seated human body to whole-body vibration vary considerably between people, but the reasons for the variability are not well understood. This study was designed to determine how the physical characteristics of people affect their apparent mass and whether inter-subject variability is influenced by the magnitude of vibration and the support of a seat backrest. The vertical apparent masses of 80 seated adults (41 males and 39 females aged 18-65) were measured at frequencies between 0.6 and 20 Hz with four backrest conditions (no backrest, upright rigid backrest, reclined rigid backrest, reclined foam backrest) and with three magnitudes of random vibration (0.5, 1.0 and 1.5 m s^-2 rms). Relationships between subject physical characteristics (age, gender, weight, and anthropometry) and subject apparent mass were investigated with multiple regression models. The strongest predictor of the modulus of the vertical apparent mass at 0.6 Hz, at resonance, and at 12 Hz was body weight, with other factors having only a marginal effect. After correction for other variables, the principal resonance frequency was most consistently associated with age and body mass index. As age increased from 18 to 65 years, the resonance frequency increased by up to 1.7 Hz, and when the body mass index was increased from 18 to 34 kg m⁻² the resonance frequency decreased by up to 1.7 Hz. These changes were greater than the 0.9-Hz increase in resonance frequency between sitting without a backrest and sitting with a reclined rigid backrest, and greater than the 1.0-Hz reduction in resonance frequency when the magnitude of vibration increased from 0.5 to 1.5 m s⁻² rms. It is concluded that the effects of age, body mass index, posture, vibration magnitude, and weight should be taken into account when defining the vertical apparent mass of the seated human body.     

Acoustic analysis of a liquefied petroleum gas-fired pulse combustor

Experimental investigation of acoustic characteristics of a Helmholtz type liquefied petroleum gas-fired pulse combustor is presented. In the experiments, the length of the tail pipe was changed from 1.9 m to 1.3 m by 10 cm intervals. Sound level measurements were taken from the exhaust side (outlet) and air flapper side (inlet) at a distance of 1 m from both sides. With decreasing lengths of the tail pipe, the sound pressure level increased. At the measurements related to the exhaust side, the maximum value of equivalent continuous sound pressure level, LEQ was 96.6 dB when the length of the tailpipe and fundamental frequency were 1.3 m and 63 Hz, respectively. Same kinds of measurements were performed at the air flapper side, but the LEQ value was stronger at the exhaust side than the one at the air flapper side. It was also observed that the effect of the type of gaseous fuel on the acoustic efficiency of the pulse combustor can be neglected when the results of the acoustic efficiencies were compared to those in the literature. In order to compare the accuracy of frequencies measured by the sound level meter, a suitable dynamic pressure transducer and a spectrum analyzer were used to perform amplitude and frequency measurements. The average deviation between the measurements performed by the sound level meter and dynamic pressure transducer was 2.4 Hz (3.8% errors) while the average deviation was 3.8 Hz (6% errors) between the sound level meter and spectrum analyzer.     

A maze structure for sound attenuation

Sonic crystals are periodic arrangement of scatterers made of material with low acoustic impedance or sound hard materials [1]. Sonic crystals have numerous applications such as green belts and sound barriers. Here we showed that a typical maze structure at children playground can attenuate noise effectively for frequencies ranging from 12.5 Hz to 20,000 Hz. The original designer for the maze structure probably does not have that in mind. The maze structure can be viewed as a sonic crystal structure with sound attenuation characteristics. We found that the maze was able to attenuate noise up to 17.9 dBA for frequency range below 1000 Hz and 23 dBA for higher frequency range up to 20,000 Hz. The maze structure was able to mitigate noise at a wide range of frequencies in addition to the center frequency (f_c$f_c$) of 478 Hz which was estimated based on the Bragg’s Law. The periodic effects of the maze was also proven by numerical studies. Our results demonstrated that the maze structure commonly found in children playgrounds was able to attenuate noise covering the whole human hearing range.     

A method for the low frequency qualification of reverberation test rooms using a validated finite element model

Low frequency behavior in small rooms is always a critical issue, but the recent extension of several standards to frequencies as low as 50 Hz opened an interesting debate in the scientific community as to which is the best (and most reliable) method to perform measurements. The present paper discusses the low frequency qualification of a typical reverberant test room in order to perform sound power measurements (carried out according to ISO 3741), by taking advantage of a finite element model of the room. Experimental measurements were first carried out in a standard reverberant chamber to demonstrate that the model provides accurate results in the range below 100 Hz. Statistical analysis of the results from measurements and simulations confirmed that, despite some small inaccuracies, the predicted results are in very good agreement with those measured both in terms of spectra and spatial distribution of the sound pressure level. Finally, the different steps of the low-frequency qualification of a reverberant test room are discussed. A selection procedure of the most suitable microphone positions is proposed, based on the results of the simulation, and, finally, on site measurements were carried out to validate the procedure.     

Acoustic array tracking performance under moderately complex environmental conditions

The direction-of-arrival (DOA) tracking performance of microphone arrays having aperture sizes ranging from 0.3 to 34 m is examined for an experiment involving a vehicle traversing a moderately complex terrain. A segment of the vehicle’s path was obscured behind a small, 6.7-m high, vegetated hill. The combination of the hill and upwind propagation created an acoustic shadow during this segment. DOA tracks were estimated with a minimum-variance distortionless response (MVDR) beamformer operating in two frequency bands: 25-60 Hz and 60-105 Hz. In the lower frequency band, array sizes between 1 and 8 m gave the best results, with DOA errors between 2° and 5°. Furthermore, in this band shadowing from the hill and wind refraction had a minimal affect on DOA error. In the higher frequency band, the acoustic shadow zone produced a distinct interval of high DOA error, with the 8-m array giving the best overall performance. Modeling of the beamforming process shows that high DOA errors corresponded to MVDR wavenumber patterns that are degraded by distortions to the propagating wavefronts. Our experimental results indicate that small acoustic arrays with apertures less than 0.3 m, operating at frequencies above 100 Hz, should be considered line of sight sensors. Given the moderate complexity of the test conditions, it is anticipated that the observed effects are likely to be present in most attempts to localize outdoor sound sources.     

Study of ultrasound stiffness imaging methods using tissue mimicking phantoms

A pilot study was carried out to investigate the performance of ultrasound stiffness imaging methods namely Ultrasound Elastography Imaging (UEI) and Acoustic Radiation Force Impulse (ARFI) Imaging. Specifically their potential for characterizing different classes of solid mass lesions was analyzed using agar based tissue mimicking phantoms. Composite tissue mimicking phantom was prepared with embedded inclusions of varying stiffness from 50 kPa to 450 kPa to represent different stages of cancer. Acoustic properties such as sound speed, attenuation coefficient and acoustic impedance were characterized by pulse echo ultrasound test at 5 MHz frequency and they are ranged from (1564 ± 88 to 1671 ± 124 m/s), (0.6915 ± 0.123 to 0.8268 ± 0.755 db cm^-1 MHz^-1) and (1.61×10⁶ ± 0.127 to 1.76 × 10⁶ ± 0.045 kg m^-2 s^-1) respectively. The elastic property Young’s Modulus of the prepared samples was measured by conducting quasi static uni axial compression test under a strain rate of 0.5 mm/min upto 10 % strain, and the values are from 50 kPa to 450 kPa for a variation of agar concentration from 1.7% to 6.6% by weight. The composite phantoms were imaged by Siemens Acuson S2000 (Siemens, Erlangen, Germany) machine using linear array transducer 9L4 at 8 MHz frequency; strain and displacement images were collected by UEI and ARFI. Shear wave velocity 4.43 ± 0.35 m/s was also measured for high modulus contrast (18 dB) inclusion and X.XX m/s was found for all other inclusions. The images were pre processed and parameters such as Contrast Transfer Efficiency and lateral image profile were computed and reported. The results indicate that both ARFI and UEI represent the abnormalities better than conventional US B mode imaging whereas UEI enhances the underlying modulus contrast into improved strain contrast. The results are corroborated with literature and also with clinical patient images.     

扬声器的特性不匹配对虚拟声像的影响

本文从理论和实验上探讨了扬声器的特性不匹配对重放虚拟声像的影响。结果表明,用两扬声器进行虚拟听觉重放时,在某些频率段和虚拟声像角度,两扬声器间很小的幅频特性差异或相频特性差异都足以对虚拟声像方向产生明显的影响。扬声器特性的差异对前方范围的声像影响较小,但对侧向范围的声像影响较大,因而两扬声器的特性不匹配也是导致虚拟听觉重放时侧向声像位置畸变的重要原因之一。而在实际应用中要特别注意两扬声器的特性匹配,或者要用信号处理的方法对两扬声器的特性进行校正。