Directional Loudspeaker Research Articles

Design of Robust Broadband Frequency-Invariant Broadside Beampatterns for the Differential Loudspeaker Array

The directional loudspeaker array has various applications due to its capability to direct sound generation towards the target listener and reduce noise pollution. Differential beamforming has recently been applied to the loudspeaker line array to produce a broadside frequency-invariant radiation pattern. However, the existing methods cannot achieve a compromise between robustness and broadband frequency-invariant beampattern preservation. This paper proposed a robust broadband differential beamforming design to allow the loudspeaker line array to radiate broadside frequency-invariant radiation patterns with robustness. Specifically, we propose a method to determine the ideal broadside differential beampattern by combining multiple criteria, namely null positions, maximizing the directivity factor, and achieving a desired beampattern with equal sidelobes. We derive the above ideal broadside differential beampattern into the target beampattern in the modal domain. We propose a robust modal matching method with Tikhonov regularization to optimize the loudspeaker weights in the modal domain. Simulations and experiments show improved frequency-invariant broadside beamforming over the 250–4k Hz frequency range compared with the existing modal matching and null-constrained methods.

Exploring variations to ASTM E336 directional sound source positioning

This paper explores variations to the procedures presented in ASTM E336 Standard Test Method for Measurement of Airborne Sound Attenuation between Rooms in Buildings. Per the Test Standard, directional sound sources must be aimed into corners away from the test partition. For source rooms horizontally adjacent to occupied spaces on all sides, multiple loudspeaker positions are needed to measure noise isolation in all directions. Measurements made in 4 office spaces are presented, each made in two ways: firstly in compliance with the Standard; and secondly, with sound sources orientated and operated differently, to explore whether using a single configuration of directional loudspeakers can produce similar results. Variations in the results are discussed, and suggestions are provided for future work.

A basic study on laboratory measurement of oblique-incidence sound transmission loss of building elements

Understanding the sound insulation performance of building elements under oblique incidence conditions is occasionally required for assessing the indoor sound environment of high-rise buildings that face roads or railways. In order to gain knowledge toward establishing a laboratory measurement method for oblique incident sound transmission loss, a measurement system was set up in a coupled reverberation room and anechoic chamber with a movable sound source. For verification of the system, a pilot measurement was carried out with a large glass pane, while the experimental set-up was also numerically simulated by FEM. In the experimental results, transmission loss was observed to depend on incidence angle in correlation with the coincidence effect, roughly obeying theory. However, it was found that the incident power estimated from the surface sound pressure on the specimen differs somewhat from that estimated using loudspeaker directivity in a free field. This discrepancy is thought due to edge diffraction from the beading around the glass pane, and this was confirmed by the numerical results. Further, this edge diffraction may also have caused the observed coincidence effect under normal incidence.

The combined effect of source directivity and binaural listening on near-field binaural speech transmission index evaluation

The binaural speech transmission index (BSTI) is mainly affected by the effect of binaural listening, background noise, and acoustic conditions in a given space, as well as source characteristics such as directivity. However, little attention has been paid to the combined effect of source directivity and binaural listening on the BSTI in near-field regions. This study thus investigates this issue using BSTI measurements in an anechoic environment to exclude the influence of acoustic conditions. Three directional loudspeakers are adopted as the sources, and the impulse responses under different radiation directions in the horizontal plane are measured in an anechoic chamber. These impulse responses are then convolved with the near-field head-related transfer functions of a KEMAR manikin to obtain the binaural impulse responses, which are further used to calculate the BSTIs. The results show that the BSTI decreases considerably with an increasing deviation of the axial direction of the source from the listener (i.e., an increasing radiation angle) due to the directivity pattern of the source, which is almost independent of the azimuth and the distance between the source and listener. Finally, the BSTI correction functions of the source radiation angles are established with third-order polynomial regression for the three sources and can be integrated into the azimuth-dependent BSTI model to yield a comprehensive model that considers the source radiation angle as well as the azimuth and distance between the source and listener.

Spherical harmonic representation of the observed directional wave front in the time domain.

Sound field reproduction algorithms require loudspeaker directivity, which is usually measured at discrete frequencies. A time domain model of loudspeaker directivity benefits broadband applications. This Letter proposes the concept of a directional wave front in the time domain, which could be linked to loudspeaker impulse responses measured on a spherical surface. The observed signal in the time domain at the boundary of a spherical region due to a propagating directional wave front is illustrated using a geometric model. Based on the geometric model, the spherical harmonic decomposition of the observed signal in the time domain is also derived.

David Blackstock and the parametric array in air

In 1971, Peter Westervelt and David Blackstock attended a Navy meeting on underwater acoustics. Following mention of rumored failed attempts to create a parametric array in air, Peter offered reasons why it should not be expected to work in air. David, sitting nearby, said to him “Bulls**t Peter!,” to which Peter responded “Alright, you prove it!” After returning to Austin, David proceeded to do just that with master’s student Mary Beth Bennett, whose experiments resulted in their 1975 JASA paper “Parametric array in air.” In the 1990s, Joseph Pompei, then a master’s student and former engineer at Bose, became interested in the airborne parametric array. His vision was to use the technology to make a practical directional loudspeaker and enable new ways to control sound. Facing similar skepticism from the audio industry, Pompei responded with a doctoral dissertation demonstrating the first high performance, low distortion parametric loudspeaker. The main technical challenges he overcame were developing correct signal processing methods and creating an ultrasonic transducer and amplifier with adequate bandwidth, power and fidelity. Now commercially successful as the “Audio Spotlight” by Holosonics, it has proved successful in digital signage, museums, retail displays, libraries, and similar places that require well localized sound.

Development of an Acoustic Spot Data Communication System Using Multiple Parametric Array Loudspeakers

In this letter, we develop an acoustic spot data communication system using multiple parametric array loudspeakers, which are directional loudspeakers and can send audio in a specific direction. In our system, data are modulated and transmitted through multiple parametric array loudspeakers, and complete data reception is possible only at a specific spot. We implemented our system using low‐cost commercially available products and a self‐developed Java program, and we used frequency‐shift keying to modulate the acoustic data communication between the sender and receiver modules. Through comparative experimental evaluations, we show that data reception at a specific spot is possible using our system. © 2021 Institute of Electrical Engineers of Japan. Published by Wiley Periodicals LLC.

Sub-array equalization technique for the parametric array loudspeaker to reduce nonlinear distortion

The parametric array loudspeaker (PAL) is a directional loudspeaker which uses the nonlinear acoustic effect, namely the parametric array, to produce an audio beam from narrow ultrasonic beams. The PAL can efficiently deliver audible information, without generating noise to the surroundings. One significant drawback of the PAL is the nonlinear distortion. Therefore, many sophisticated methods have been proposed to preprocess the input signal of the PAL. However, those methods usually request a flat frequency response of the ultrasonic transducer array (UTA). In the past, equalization has been tried out for the whole UTA, but the performance was sometimes not satisfactory due to the inconsistent productions of ultrasonic transducers. This paper proposes to group the ultrasonic transducers by their impedances. Several sub-arrays are thereafter formed and equalized individually. The comparison results demonstrate that the propose sub-array equalization technique can suppress the nonlinear distortion of the PAL more effectively than the previous method.

Simulation study on characteristics of acoustic metamaterials based on Mie and Helmholtz resonance for low-frequency acoustic wave control

An increasing number of applications require a low-frequency acoustic source to achieve a high sound pressure level (SPL) in the far field; however, controlling low-frequency waves remains a challenge. Recently, phononic crystals have provided a new way to control acoustic waves. For example, the Mie resonant structure, which can reflect low-frequency acoustic waves effectively with its subwavelength size, has been used to realize an acoustic collimated beam. In this work, on the basis of the Mie resonant structure, we designed a Mie-Helmholtz structure (M-H structure) that can use the high sound pressure inside the central cavity to further enhance the far-field SPL and unidirectional directivity. The developed design consists of adding two shells on both sides of the maze-like structure and creating a hole in the middle of the upper shell. We used modal analysis, equivalent medium method to study the structure and obtained lumped parameter model, finding that the M-H structure has physical properties in the XY plane similar to those of the Mie structure while showing a Helmholtz amplification effect in the Z direction. To explore the effect of the new structure on realizing the directivity of low-frequency waves, we also compared the sound field between M-H structure and M-structure. The simulation results showed 8.4% enhancement of the far-field sound intensity and 61.9% enhancement of the directivity index (DI) in main lobe direction. We also provided an example of how this design can be applied to improve low-frequency loudspeaker directivity by adding 4M-H structures and found that the radiation power was restrained in the XY plane and showed better unidirectional directivity (DI ⩾4 dB) in the Z direction. The location of M-H structure for better unidirectional directivity (DI = 9.1 dB) have also been investigated. Finally, we studied the influence of different sizes on the resonance frequency and the SPL gain at the middle hole to explore a guidance for design.

Rerouting Hearing Aid Systems for Overcoming Simulated Unilateral Hearing in Dynamic Listening Situations.

Unilateral hearing loss increases the risk of academic and behavioral challenges for school-aged children. Previous research suggests that remote microphone (RM) systems offer the most consistent benefits for children with unilateral hearing loss in classroom environments relative to other nonsurgical interventions. However, generalizability of previous laboratory work is limited because of the specific listening situations evaluated, which often included speech and noise signals originating from the side. In addition, early studies focused on speech recognition tasks requiring limited cognitive engagement. However, those laboratory conditions do not reflect characteristics of contemporary classrooms, which are cognitively demanding and typically include multiple talkers of interest in relatively diffuse background noise. The purpose of this study was to evaluate the potential effects of rerouting amplification systems, specifically a RM system and a contralateral routing of signal (CROS) system, on speech recognition and comprehension of school-age children in a laboratory environment designed to emulate the dynamic characteristics of contemporary classrooms. It was expected that listeners would benefit from the CROS system when the head shadow limits audibility (e.g., monaural indirect listening). It was also expected that listeners would benefit from the RM system only when the RM was near the talker of interest. Twenty-one children (10 to 14 years, M = 11.86) with normal hearing participated in laboratory tests of speech recognition and comprehension. Unilateral hearing loss was simulated by presenting speech-shaped masking noise to one ear via an insert earphone. Speech stimuli were presented from 1 of 4 loudspeakers located at either 0°, +45°, -90°, and -135° or 0°, -45°, +90°, and +135°. Cafeteria noise was presented from separate loudspeakers surrounding the listener. Participants repeated sentences (sentence recognition) and also answered questions after listening to an unfamiliar story (comprehension). They were tested unaided, with a RM system (microphone near the front loudspeaker), and with a CROS system (ear-level microphone on the ear with simulated hearing loss). Relative to unaided listening, both rerouting systems reduced sentence recognition performance for most signals originating near the ear with normal hearing (monaural direct loudspeakers). Only the RM system improved speech recognition for midline signals, which were near the RM. Only the CROS system significantly improved speech recognition for signals originating near the ear with simulated hearing loss (monaural indirect loudspeakers). Although the benefits were generally small (approximately 6.5 percentage points), the CROS system also improved comprehension scores, which reflect overall listening across all four loudspeakers. Conversely, the RM system did not improve comprehension scores relative to unaided listening. Benefits of the CROS system in this study were small, specific to situations where speech is directed toward the ear with hearing loss, and relative only to a RM system utilizing one microphone. Although future study is warranted to evaluate the generalizability of the findings, the data demonstrate both CROS and RM systems are nonsurgical interventions that have the potential to improve speech recognition and comprehension for children with limited useable unilateral hearing in dynamic, noisy classroom situations.

Contralateral Routing of Signal Systems Can Improve Speech Recognition and Comprehension in Dynamic Classrooms.

Objective The purpose of this study was to evaluate the effects of hearing aid-based rerouting systems (remote microphone [RM] and contralateral routing of signals [CROS]) on speech recognition and comprehension for children with limited usable hearing unilaterally. A secondary purpose was to evaluate students' perceptions of CROS benefits in classrooms. Method Twenty children aged 10-16 years with limited useable hearing in one ear completed tasks of sentence recognition and comprehension in a laboratory. For both tasks, speech was presented from one of four loudspeakers in an interleaved fashion. Speech loudspeakers were either midline, monaural direct, or monaural indirect, and noise loudspeakers surrounded the participant. Throughout testing, the RM was always near the midline loudspeaker. Six established users of CROS systems completed a newly developed questionnaire that queried experiences in diverse listening situations. Results There were no effects of RM or CROS use on performance for speech presented from front or monaural direct loudspeakers. However, for monaural indirect loudspeakers, CROS improved sentence recognition and RM impaired recognition. In the comprehension task, CROS improved comprehension by 11 rationalized arcsine units, but RM did not affect comprehension. Questionnaire results demonstrated that students report CROS benefits for talkers in the front and from the side, but not for situations requiring localization. Conclusions The results support CROS benefits without CROS disadvantages in a laboratory environment that reflects a dynamic classroom. Thus, CROS systems have the potential to improve hearing in contemporary classrooms for students, especially if there is only a single microphone.

Auralization of High-Order Directional Sources from First-Order RIR Measurements

Can auralization of a highly directional source in a room succeed if it employs a room impulse response (RIR) measurement or simulation relying on a first-order directional source, only? This contribution presents model and evaluation of a source-and-receiver-directional Ambisonics RIR capture and processing approach (SRD ARIR) based on a small set of responses from a first-order source to a first-order receiver. To enhance the directional resolution, we extend the Ambisonic spatial decomposition method (ASDM) to upscale the first-order resolution of both source and receiver to higher orders. To evaluate the method, a listening experiment was conducted based on first-order SRD-ARIR measurements, into which the higher-order directivity of icosahedral loudspeaker’s (IKO) was inserted as directional source of well-studied perceptual effects. The results show how the proposed method performs and compares to alternative rendering methods based on measurements taken in the same acoustic environment, e.g., multiple-orientation binaural room impulse responses (MOBRIRs) from the physical IKO to the KU-100 dummy head, or higher-order SRD ARIRs from IKO to em32 Eigenmike. For optimal externalization, our experiments exploit the benefits of virtual reality, using a highly realistic visualization on head-mounted-display, and a user interface to report localization by placing interactive visual objects in the virtual space.

Parlante superdireccional de matriz paramétrica

Este trabajo trata sobre el diseño y construcción de un altoparlante direccional, también conocido como parlante de matriz paramétrica (PAL). El funcionamiento del mismo se basa en los principios de un fenómeno denominado autodemodulación que sucede cuando haces de alta frecuencia interactúan en un medio viscoso, debido a las propiedades no lineales del mismo. El sonido obtenido luego de la autodemodulación se forma en el haz de la señal portadora ultrasónica propagándose más allá de la distancia en la que la portadora ha sido suficientemente atenuada. Esto genera una señal audible focalizada y por ende un patrón polar direccional de ángulo angosto. En el presente trabajo se evalúan dos métodos de modulación: en amplitud con doble banda lateral (AM-DBL) y por variación de ancho de pulso (PWM). Se analiza experimentalmente su efecto en la direccionalidad, nivel de presión sonora y distorsión armónica.

Near Direct and Far Reverberant Field Response of Direct Radi ator Loudspeakers

Typical audience seating arrangements in rooms and auditoria warrant reinvestigation of the direct radiator speaker response in the near-direct and far-reverberant fields, as the response data provided by the manufacturer is always ideal and does not account for the effect of those fields. The speaker response characteristics of a variety of direct radiator loudspeakers ranging from the conventional squawker to the full range radiator have been investigated in these fields. The speaker response is investigated in the 50 -10 kHz frequency range, by measuring the A-weighted SPL (sound pressure level) in the near-direct and far-reverberant fields, using an acoustic analyzer. The field-specific characteristic for each of the radiators is determined by fitting the SPL data obtained to an appropriate polynomial. The coefficients obtained thereby, allow an objective field-specific study amongst radiators. When a set of direct radiator loudspeakers is available, it is necessary to configure their application, depending upon the optimum sound quality required for a given enclosure, in near-direct field and far-reverberant field. The outcome of this work assists one to configure the best radiator ensemble for a given enclosure, despite placement constraints.

Noise predictions of sound systems using system data and complex summation

As the number and size of amplified outdoor events in urban areas is increasing, so are the challenges with accompanied noise immissions and therefore the need for accurate noise predictions. An advanced method for predictions of sound systems is introduced, that uses specific system data and applies complex summation. The system data, including all acoustically relevant characteristics, such as positioning, orientation and directivity of loudspeakers, delays and electronic filters, are simply imported with a system design file already in use by the system provider (d&b’s ArrayCalc simulation software). This procedure eliminates any inaccuracies and inefficiencies because it spares the process of re-modeling a sound system in the noise prediction software and ensures predictions using the actual system design. Also, and with increasing importance, advanced electronic filtering (e.g., IIR, FIR) is automatically included. Environmental noise modelling software usually does not consider complex summation between sources because they are assumed to be un-correlated with one another (as in traffic and industry). However, in sound systems, the signals sent to the loudspeakers are mostly correlated. Furthermore, modern line-array systems use coherence effects to further tailor directivity. Therefore, the sound propagation calculation was extended for complex summation and the method was implemented in SoundPLAN software and in NoizCalc (d&b audiotechnik).

Sound Straight Ahead: Parametric Speakers in Two Soundscape Installations

This article concerns the realization of two sound installations that use a parametric (directional) loudspeaker as a central element of their working systems. The Soundhouse is a rotating structure that projects sounds in a directional way. El bosque y las sombras is instead an interactive multimedia setting where the visitors can trigger sounds and affect a video projection by moving in an enclosed, soundproofed space. A short introduction to parametric loudspeakers is provided; then the concept and the construction aspects of both installations are presented. Finally, an overall aesthetic assessment of the use of parametric loudspeakers in the above works is given.

Low Cost Omnidirectional Sound Source Utilizing a Common Directional Loudspeaker for Impulse Response Measurements

Alternative low-cost sources (e.g., balloons, gun fires) are used for impulse response measurements when a dodecahedron speaker is not available. This study sets to explore the applicability of a method utilizing a common directional loudspeaker as a sound source. For this purpose measurements were performed in three spaces with three different common directional loudspeakers. Different placements of the loudspeakers were performed (twelve positions similar to the twelve positions of the faces of a dodecahedron speaker, different rotations of the loudspeakers for a total sum of twenty six and fourteen positions). The impulse responses obtained were added up creating a single impulse response for each case. Comparisons of the acoustic parameters measured with the proposed method and with a dodecahedron speaker are presented and suggest the expected mean absolute error and standard deviation for similar measurements. Reverberation time measurements show a mean absolute error of less than 0.08 s, as compared with measurements with a dodecahedron speaker. The proposed method can be the primary method for measuring impulse responses when a dodecahedron speaker is not available. Suggested improvements may lead to better omnidirectionality as compared with a dodecahedron loudspeaker, and set the method applicable to be utilized for auralization purposes.

A “composite” stepped plate for use in highly directional parametric array loudspeakers

It has been shown that stepped plate radiators with appropriately designed steps can be used in highly directional parametric array (PA) loudspeakers [Hwang and Moon, 170th Meeting of the Acoustical Society of America, 138(3), 1801(2015)]. To improve the performance of the PA loudspeaker, a relatively thin plate with steps needs to be adopted for its higher power efficiency. However, the natural frequencies of the plate and the corresponding mode shapes depend on the mass of the steps. Hence, it can be difficult to design a stepped plate suitable for use in PA loudspeakers. In this paper, we propose a novel plate design, with steps composed of a stiff, lightweight, polymer material. The steps were fabricated with a thickness of 2.3 mm from a polymer material with an elastic modulus of 1 GPa, a Poisson ratio of 0.23, and a mass density of 540 kg/m3 on the plate with a thickness of 1 mm. Experiments and simulations were carried out to determine the acoustic characteristics of our fabricated “composite” stepped plate. [Work supported by NRF 2016R1E1A2A02945515).]

Visualizing sound directivity via smartphone sensors

We present a fast, simple method for automated data acquisition and visualization of sound directivity suitable for use with introductory acoustics education, made convenient and accessible via a new free smartphone app, “Polar Pattern Plotter.” The app synchronizes measurements of sound volume with the phone's angular orientation, obtained from either compass, gyroscope, or accelerometer sensors, and produces a graph and exportable data file. It is generalizable to various sound sources and receivers via the use of an input-jack-adaptor to supplant the smartphone's (omnidirectional) microphone. We show results obtained in a free-field environment for loudspeaker directivity, microphone polar patterns, and two-speaker interference effects. Results provide both a visual and quantitative representation of sound fields and device responses, useful for introductory acoustics experiments and demonstrations.

Binaural active noise control using parametric array loudspeakers

This paper reports the binaural active noise control (ANC) system developed to deal with factory noise. The control points are located in the vicinity of the left and right ears of a worker sitting along the production line. Due to the complicated safety requirements in the factory, secondary sources and error microphones are not allowed to be placed near the worker. Therefore, the proposed ANC system employs the feedforward structure and adopts the parametric array loudspeakers (PALs) as the secondary sources. The PAL is a type of directional loudspeaker that generates a much narrower sound field as compared to the conventional loudspeaker. Once the proposed ANC system has been trained offline, the error microphones can be removed. The performance of the binaural ANC system is successfully demonstrated based on a digital signal processor (DSP) implementation.