Loudspeaker System Research Articles

A feedforward and feedback composite active noise reduction headset based on inverse filter frequency equalization and its DSP system implementation

The principle of active noise reduction headset is to reduce the original noise by generating inverted noise through the loudspeaker. The loudspeaker system will introduce spectrum distortion inevitably. This distortion will increase the complexity of the secondary channel model, which adversely affect identification of the primary channel in the real-time active noise reduction system. To improve the inherent frequency spectrum distortion of the speaker system, this paper proposes a new frequency equalization method based on inverse filtering. This method is applied to a feedforward and feedback compound active noise reduction earphone system, and a new active noise reduction headphone system is obtained. In order to improve the noise reduction effect, the pollution caused by the secondary sound source to the reference signal is compensated in the active noise reduction control scheme of this paper. Simulation and real-time noise reduction experiments based on DSP solution show that this new active noise reduction control system can achieve better noise reduction effects.

REPAIR OF LOUDSPEAKERS AS A SUPPORT FOR WORSHIP ACTIVITIES AT THE AL-MADINAH AL-MUNAWWAROH PERUM MOSQUE. SUNRISE GARDEN BUKIT PUTRA CILEUNGSI WEST JAVA

Mosque loudspeakers or mosque speakers are an electronic device used to increase the sound intensity in mosques. Usually, each mosque has two loudspeaker systems, namely inside and outside. External loudspeakers are generally used for adhan and iqomah, while for others such as prayers, lectures and tadarus only use internal speakers. However, in several areas, a number of mosques, regardless of their neighbors who live nearby, also use their external loudspeakers for various activities other than the call to prayer and iqomah, which creates problems in society. Scientists, including those from Harvard University, found that besides causing hearing loss, noise pollution can also reduce intelligence, memory, mental health, lead to cardiovascular disorders (heart disease and stroke), type 2 diabetes mellitus, etc

Efficient Realization for Third-Order Volterra Filter Based on Singular Value Decomposition

Nonlinear distortion in loudspeaker systems degrades sound quality and must be properly compensated for by linearization techniques. One technique to reduce nonlinear distortion is to use a Volterra Filter, which approximates the nonlinearity of the target loudspeaker using the Volterra series expansion. In general, the Volterra Filter is computationally very expensive, and the amount of computation needs to be reduced for real-time processing. In this paper, we propose an efficient implementation of the third-order Volterra filter based on singular value decomposition. The proposed method determines the necessary coefficients based on the symmetry of the third-order Volterra filter and applies singular value decomposition to them. In the filter structure consisting of singular values and their corresponding singular vector, the computational complexity of the third-order Volterra filter can be reduced by eliminating the part of the filter with small singular values. By focusing on the magnitude of the singular values, the proposed method can improve the computational efficiency of the third-order Volterra filter without decreasing its approximation accuracy. Simulation results show that the proposed method can improve the computational efficiency by 60% while maintaining the nonlinear distortion compensation performance of the micro-speaker for smartphones by about 8 dB.

Sound intensity and directivity measurements for a loudspeaker canopy in a restrictive noise environment

It is common to have noise complaints from outdoor event and entertainment venues in urban and non-urban environments due to the loudness of the reinforcement system, rhythmic nature of the music, acoustical spectrum, and the geometric spreading of said system. Complaints about noise from events, bars, and restaurants are on the rise. Increasing traffic, construction, and commercial activities tend to increase the ambient noise level in urban environments, creating noise creep. The pandemic restrictions helped reduce the ambient levels in the urban and non-urban areas. However, after the reopening, more and more noise complaints started to be issued since residents were more sensitive to noise pollution. This paper presents a practical application of the source directivity for a loudspeaker system. The loudspeaker canopy is designed to produce a radiation pattern directing the main beam lobe to the front of the array and reduce the noise propagation to other directions. This device can be implemented in entertainment and outdoor events (such as wedding parties) to reduce the noise impact on sensitive receivers and meet the jurisdictional noise regulations. Real-world acoustic intensity measurements were made to obtain the system's sound power and assess the noise impact at distances up to 35 feet away from the source. The viability of using the system for outdoor activities within locations restricted by the City's ordinances is discussed.

Two Sound Field Control Methods Based on Particle Velocity

In recent years, a variety of sound field control methods have been proposed for the generation of separated sound regions. Different algorithms control the physical properties of the generated sound field to different degrees. The existing methods mainly focus on sound pressure restoration and its related improvement. When the loudspeaker array is non-uniformly placed, the reconstruction system is not stable enough. To solve this problem, this paper proposes two sound field control methods related to particle velocity. The first method regulates the reconstruction error of particle velocity in the bright zone and the square of particle velocity in the dark zone; the second method regulates the reconstruction error of sound pressure and particle velocity in the bright zone and the square of sound pressure and particle velocity in the dark zone. Five channel and twenty-two channel non-uniform loudspeaker systems were used for two-dimensional and three-dimensional computer simulation testing. Experimental results show that the two proposed methods have better tradeoffs in terms of acoustic contrast, reproduction error and array effort than traditional methods, especially the second proposed method. In the two-dimensional experiment, the maximum reductions of the average array efforts generated by the proposed methods were about 10 dB and 11 dB compared with the average array efforts generated by two traditional methods. In the three-dimensional experiment, the maximum reductions of the average array efforts generated by the proposed methods were about 8 dB and 2 dB compared with the average array efforts generated by two traditional methods. The smaller the array effort, the more stable the loudspeaker system. Therefore, the reconstruction systems produced by the proposed methods are more stable than those produced by the traditional methods.

Influence of microphone characteristics on demodulated sound measurement in near field of parametric loudspeaker

This study evaluates the accuracy of demodulated sound measurements using a condenser microphone in the near field of a parametric loudspeaker system. Microphones with different sensitivities placed at incidence angles of 0° and 90° were used to measure demodulation frequency components without special acoustic filters. The measured components were compared with theoretical predictions. The results show that the measured sound pressure using microphones placed at 0° was up to several tens of decibels larger than the theoretical predictions and significantly inaccurate in the near field. This was due to the nonlinear response of the microphone, which had high sensitivity at primary sound frequencies, inducing spurious signals. This result suggests that using a microphone with low sensitivity at primary sound frequencies placed at an appropriate angle that reduces sensitivity improves parametric sound measurement accuracy.

On the physical origin of the electro-mechano-acoustical analogy.

The electro-mechano-acoustical (EMA) analogy, introduced in the early 1900s, allows for the modeling of mechanical, acoustical, and EMA devices and systems using schematics of basic electrical elements, making possible the application of the well-established electrical network theory for studying, designing, and characterizing complex systems. By means of this elegant modeling, the behavior of intrinsically heterogeneous devices, such as microphones and loudspeaker drivers and systems, can be explained in each of the three domains and the mutual interactions between the electrical, mechanical, and acoustical parts can be precisely predicted. However, an open issue remains still unsolved: in all publications and textbooks, the EMA analogy has been always introduced only from the formal similarity of the equations describing the physical laws that regulate the basic EMA elements and a lack of insight into its physical fundamentals can be remarked. In this paper, an investigation on the existence of physical origins of the correspondences and similarities between the quantities and the laws describing the elements of electrical, mechanical, and acoustical domains is proposed and discussed. Historical research on the birth of EMA analogy theory with the fundamental contributions on its development given by the scientists and engineers is presented as well.

Subjective responses between real impact sound and rubber ball im-pact sound

By the recent COVID-19 situation, people stay more time in their home and abatements on noise between neighbouring units are increasing. Heavy/soft impact sound is one of the major noise sources in high-rise apartment buildings. Standardized heavy/soft impact source is known for having the most similar physical and subjective characteristics with real impact sound such as a child running, jumping and an adult walking. The single number quantity on the rubber ball was standardized. A classification scheme for rubber ball impact sound needs to be standardized. Several studies on subjective responses were conducted on rubber ball impact sound in various situations. In this study, subjective responses on the rubber impact sound and real impact sound were compared. The subjective experiment was conducted in the listening chamber which is furnished similarly to the typical living room of Korean apartment buildings. In the experiment, rubber ball impact sounds recorded in the real apartment building and real impact sound recorded in the mock-up building were presented through a sub-woofer and multi-channel loudspeaker system. Subjective responses were collected with an 11 points SD scale.

Developing The Firm Mosque Model As A Public Health Institution Through The Implementation Of Psychoacoustic On Loudspeaker Systems

Voices originating from the mosque, whether they are recitation, incidental activities or daily routine prayers, are still often interpreted as a source of noise. Not only because of the improper arrangement of the loudspeakers, but also because the mosque's recitals or takmirs generally do not understand the psychoacoustic aspect. Psychoacoustic is the study of the effect of sound exposure on the psychological aspects of the listeners. The use of psychoacoustic science could potentially change mosque sounds into therapeutic voices. The voices of recitation-based therapists have been encountered and researched, although in a limited scope. Building a mosque sound management and control system that is consistent within a psychoacoustic framework can transform potential noise into a natural sound that builds public health that runs automatically. The achievement of this hypothesis can be done from the application of a research and / or community service activity by educational institutions or related agencies.

A computationally efficient hybrid 2D\u20133D subwoofer model

A subwoofer generates the lowest frequency range in loudspeaker systems. Subwoofers are used in audio systems for live concerts, movie theatres, home theatres, gaming consoles, cars, etc. During the last decades, numerical simulations have emerged as a cost- and time-efficient complement to traditional experiments in the design process of different products. The aim of this study is to reduce the computational time of simulating the average response for a given subwoofer design. To this end, we propose a hybrid 2D–3D model that reduces the computational time significantly compared to a full 3D model. The hybrid model describes the interaction between different subwoofer components as interacting modules whose acoustic properties can partly be pre-computed. This allows us to efficiently compute the performance of different subwoofer design layouts. The results of the hybrid model are validated against both a lumped element model and a full 3D model over a frequency band of interest. The hybrid model is found to be both accurate and computationally efficient.

Hospital entrance smoking is reduced by broadcasting recorded antitobacco messages from Australian primary school children over entrance public address system.

Smoking near hospital entrances occurs frequently despite smoke-free policies, resulting in multiple issues including second-hand smoke exposure (SHS) to vulnerable populations. Primary school children were engaged through their health curriculum to produce antismoking audio recordings for broadcast over a hospital entrance loudspeaker system to determine if this reduced smoking. Students produced original recordings against hospital grounds smoking during class workshops, from which a collection (n=16) was selected. Episodes of entrance smoking and total entrance traffic were recorded using security camera infrastructure over a 5-week period. A computer-controlled entrance loudspeaker played a message which was followed by silence until a new (different) message was played. Intensity of messaging was moderate in week 3 (every 5minutes), increasing to high in week 4 (3 minutely) and compared to no messages (weeks 1-2 preintervention) and week 5 (postintervention). Smokers presented 316 times, smoking 523 cigarettes over 155hours of observation (patients 70.6%, visitors 29.4%). SHS exposure was high given 172 others used the entrance/hour. Smoking was highest in the preintervention period (weeks 1-2), median five cigarettes/hr [IQR,3-7 (min=0, max=12)], falling to four cigarettes/hr during 5-minutely broadcasts [IQR,2-5 (min=1, max=14)] (P=.06), but zero cigarettes/hr during 3-minutely broadcasts [IQR,0-1(min=0, max=7)] (P<.0001). Postintervention (no broadcasts), smoking increased from zero to 1 cigarette/hr [IQR,0-3 (min=0, max=5)]; (P=.052). Nonsmoker movements did not change significantly between each period. Intensive (3minutely) broadcasting of short antismoking messages significantly reduced hospital entrance smoking. SO WHAT?: Health services can positively interact with the health curriculum of primary schools against tobacco use while developing low-cost strategies to effectively deter entrance smoking.

Large-scale outdoor sound field control.

The feasibility and the performance of controlling low frequency sound of loudspeaker systems under varying atmospheric conditions is examined experimentally. In the experiment, a control subwoofer array is canceling the sound of a primary subwoofer array over long distances (∼100 m) and in large areas (∼320 m2) using the pressure-matching method. To avoid the measurement of the sound field over the entire control area, a sound propagation model is introduced that is fitted in situ to model the radiation properties of the loudspeakers and the variation of the speed of sound. The results show that the control system reduces the sound pressure levels by up to 15-20 dB over the subwoofers' frequency range. However, the reduction can vary considerably depending on the specific atmospheric condition. The model-based approach reduces the number of required measurements and achieves similar reduction performance to the control based on direct measurements with considerably fewer microphone locations while also being more robust. Additionally, the sound propagation model enables the reduction of acoustic energy in virtual control zones that are far away from the microphone location. The investigated methodology has a direct application in the mitigation of sound from outdoor concerts.

Content-sensitive warped filter equalization for loudspeaker systems

The technique of frequency-warped filtering has been recommended as a promising solution to the problem of loudspeaker equalization. According to the characteristics of audio signal, this paper puts forward a method to improve the warping effect and thus the overall equalization effect, and the effectiveness of the proposed method is validated via computer simulations.

Designing Audio Equalization Filters by Deep Neural Networks

Audio equalization is an active research topic aiming at improving the audio quality of a loudspeaker system by correcting the overall frequency response using linear filters. The estimation of their coefficients is not an easy task, especially in binaural and multipoint scenarios, due to the contribution of multiple impulse responses to each listening point. This paper presents a deep learning approach for tuning filter coefficients employing three different neural networks architectures—the Multilayer Perceptron, the Convolutional Neural Network, and the Convolutional Autoencoder. Suitable loss functions are proposed for each architecture, and are formulated in terms of spectral Euclidean distance. The experiments were conducted in the automotive scenario, considering several loudspeakers and microphones. The obtained results show that deep learning techniques give superior performance compared to baseline methods, achieving almost flat magnitude frequency response.

Auditory augmented process monitoring for cyber physical production systems

We describe two proof-of-concept approaches on the sonification of estimated operation states and conditions focusing on two scenarios: a laboratory setup of a manipulated 3D printer and an industrial setup focusing on the operations of a punching machine. The results of these studies form the basis for the development of an “intelligent” noise protection headphone as part of Cyber Physical Production Systems which provides auditorily augmented information to machine operators and enables radio communication between them. Further application areas are implementations in control rooms (equipped with multi-channel loudspeaker systems) and utilization for training purposes. As a first proof-of-concept, the data stream of error probability estimations regarding partly manipulated 3D printing processes were mapped to three sonification models, providing evidence about momentary operation states. The neural network applied indicates a high accuracy (> 93%) of the error estimation distinguishing between normal and manipulated operation states. None of the manipulated states could be identified by listening. An auditory augmentation, or sonification of these error estimations, provides a considerable benefit to process monitoring. For a second proof-of-concept, setup operations of a punching machine were recorded. Since all operations were apparently flawlessly executed, and there were no errors to be reported, we focused on the identification of operation phases. Each phase of a punching process could be algorithmically distinguished at an estimated probability rate of > 94%. In the auditory display, these phases were represented by different instrumentations of a musical piece in order to allow users to differentiate between operations auditorily.

Preliminary experimental evaluation of a novel loudspeaker featuring magnetorheological fluid surround absorber

&lt;span&gt;A novel design of magnetorheological fluids (MRF) based surround device in a loudspeaker system was studied in this article. The main objective of this research is to design a new surround device of the loudspeaker that can be easily controlled its damping. Therefore, it was predicted that the audio pressure level on the loudspeaker could be easily manipulated at a different sound source by applying a certain magnetic field. This function could not be reached using one conventional speaker system. Firstly, a set of an electromagnetic device containing MRF was designed to replace the conventional rubber surround. The magnetic circuit was then evaluated using the finite element method magnetics to study the flux distribution in the MRF area. The current was varied from 0.25 to 0.75 A by an interval of 0.25 A. The magnetic flux resulted from the simulation was then logged and used as the based value for predicting the change of shear yield stress. The base properties of the shear yield stress of the MRF against the magnetic flux was obtained from previous experimental result. Therefore, it was hopefully the prediction could be closed to the real system. Based on the simulation result, the shear yield stress varied from 43 to 49 Mpa or about 15 % increment. &lt;/span&gt;&lt;span lang="IN"&gt;A simple experimental work was carried out. By applying particular direct current into the coil, the sound quality generated by the loudspeaker shows different values&lt;/span&gt;&lt;span&gt;.&lt;/span&gt;&lt;span lang="IN"&gt; Based on the preliminary experiment, the level of decibel decreased about 3 dB as the application of magnetic fields. The idea has been proven in this preliminary experimental evaluation.&lt;/span&gt;

A Critical Step to Using a Parametric Array Loudspeaker in Mobile Devices.

A parametric array (PA) loudspeaker is a highly directional audio source that might grant one's convenience if it is used with mobile devices. However, conventional PA loudspeakers is almost impossible to apply in mobile devices using a battery because of the large power consumption and large device size. In this study, a PA loudspeaker system (PALS) was fabricated and evaluated to show that those difficulties could be overcome to apply it to mobile devices. In order to construct a PALS for demonstration, a power amplifier and signal-processing unit should also be properly designed and built. The PA source transducer should also be designed and built for a mobile device application. These components were integrated into a single PALS. The PALS generated a 125-dB primary wave and 62 dB of a different frequency wave (DFW) through the PA at 0.45 m in a 3 m × 3 m × 2 m semi-anechoic chamber. We confirmed that the half-power bandwidth (HPBW) formed a 6° beam at 83 kHz of DFW and 90 kHz of the primary wave (PW), and the HPBW formed a 7.3° beam at 5 kHz of DFW and a 7.1° beam at 10 kHz of DFW, respectively. Lastly, the power required was 6.65 W without a matching circuit, and 3.25 W with such a circuit.

Virtual Loudspeaker Effect of Graphene-Based Hybrid Material To Improve Low-Frequency Acoustic Performance.

Closed-box loudspeaker systems (CBLSSs) are compact and simple air-suspension loudspeaker systems, and their low-frequency responses are determined by two fundamental parameters: resonance frequency and total damping. Recently, electronic devices have come to require more compact designs, so the volumes of loudspeaker should be reduced. However, a small loudspeaker cannot retain sufficient acoustic space, resulting in poor low-frequency acoustic performance. Herein, we investigated acoustic characterization of the CBLSS with different filling materials such as thermally expanded graphene oxide (TEGO), activated carbon, graphene platelets, and melamine foam (MF). Upon the powder-based test, the resonance frequency of the loudspeaker decreased and resulted in a volume increasing effect inside of the loudspeaker. The TEGO shows almost double volume increase rate, compared to other particle-based filling materials. Employing hybrid filling material that consists of TEGO in an MF cage (TEGO@MF), the volume increase rate of the novel loudspeaker was over 24% at 300 cc. Because of the high adsorptive characteristics and thermal properties of TEGO, the acoustic performance in the low-frequency domain was clearly enhanced, despite the reduced mass loading. Furthermore, these properties were observed to be highly effective for enhancing the low-frequency acoustic performance of the larger loudspeaker, achieving a volume increase rate of 49.5% in a 700 cc enclosure.

An electroacoustic method for measuring airflow resistivity of porous sound-absorbing materials

In this paper, a method for measuring the airflow resistivity of air-saturated porous sound-absorbing materials is presented. The method is based on a modification of the previous device developed by Dragonetti et al. The approach used in the present work involves a cavity and a Helmholtz resonator that are coupled through a loudspeaker so that the complete system behaves as a fourth-order symmetrical band-pass loudspeaker system. After a straightforward calibration, the airflow resistivity of a material sample is indirectly estimated from the direct measurement of the total electric impedance at the loudspeaker connection terminals. In this way, the use of microphones is not necessary, which makes its implementation very simple and inexpensive. Experimental results obtained with the present method agree well with those obtained through a standardized method as long as the values of the material’s airflow resistance are not too high.

Coping with Cage: On Organ2/ASLSP, Listening, and Music-Making

Coping with Cage:On Organ2/ASLSP, Listening, and Music-Making Hans Fidom (bio) During the evening session of the ASAP symposium in Amsterdam on May 25, 2018, I played the first line of John Cage's composition ORGAN2/ASLSP (As SLow aS Possible) on the little organ in the Kerkzaal, a room on the top floor of the University's main building, which was originally built to serve as a church hall. My companion was sound artist Mirjam Meerholz, who contributed a soundscape produced by a seven-part loudspeaker system, distributed across the hall and on the adjoining roof terrace. It made us think. How to cope with Cage in the context of a conference? MARYLAND In 1985, John Cage composed ASLSP for a piano competition at the University of Maryland. The piece has eight parts, one of which the pianist should choose to omit, and another one to repeat. One could imagine the pianist making a tone appear only when the former one has faded away completely, as that would make the music sound "as slow as possible." One could even expect such an approach to be to Cage's liking, as he himself "translated" ASLSP into "as slow as possible," but he actually gave no clues as to how it should be played; in fact, his instructions include the remark that "neither tempo nor dynamics have been indicated." To Thomas Moore, who premiered the piece and wanted to know how long the performance preferably would take, Cage responded, rather Zen-like: "If each section took one minute to play, then the piece would last eight minutes."1 In 1987, following a suggestion by German organist Gerd Zacher, Cage made the organ version, ORGAN2/ASLSP. Zacher premiered the piece the same year in Metz, France. It took him about twenty-nine minutes to play it. HALBERSTADT On September 5, 2001, a very special performance of ORGAN2/ASLSP started in the St. Burchardi church in Halberstadt, Germany: it is planned to last until at least the year 2640.2 Actually, the performance is the only function of the church. Gutted completely, having been abandoned for decades, it appears to be the perfect place for the likewise unpretentious yet significant music. The plan to have ORGAN2/ASLSP sound for centuries was born in 1998 in Trossingen, Germany. Following the First International Week of New Organ Music that had taken place a year earlier in the same town, organists including Christoph Bossert, Hans-Ola Ericsson, and Karin Gastell came together with organ builders including Gerald Woehl to think about time in music. Obviously, they [End Page 496] were inspired by the piece's title. They were aware that its performance might not need to be longer than it would take to say "lsp," as Cage had related ASLSP to that word in the final chapter of James Joyce's Finnegans Wake: "Soft morning, city! Lsp!"3 Yet taking the endlessness of organ tones instead to determine how slow "as slow as possible" could be fascinated them more. In principle, this idea is untenable: playing as slow as possible on an organ would mean that the music would never get beyond the first tones, as they would only die when the wind flow to the pipes producing them stopped—which it won't, provided the organ is maintained well. Hence, the Trossingen group envisioned Cage's music sounding so slowly that experiencing it would evoke a sensation of endlessness. A decisive practical step was taken by identifying Halberstadt as the birthplace of modern music: the little town once housed in its cathedral the first large organ equipped with keyboards providing twelve tones per octave.4 The fact that the organ was built in 1361 inspired the Trossingen group to a literally far-reaching idea: the time between 1361 and 2000 spans 639 years, so what if the performance would start in 2000 and last until 2639? Once again, and as always, Cage inspired keeping opposite options open; just like he mentions in his preliminary note to his famous piece 4′33″ regarding the length of its performance ("the work may … last any length of time"), ORGAN2/ASLSP...