Zone Of Quiet Research Articles

Performance evaluation of an active headrest system using a filtered-x least mean square/fourth algorithm with virtual sensing.

In an active headrest system virtual sensing tends to transfer the spatial zone of quiet from the residual error microphone to the ear canal. An attempt has been made in this paper to develop an auxiliary filter based virtual sensing scheme integrated with the filtered-x least mean square/fourth algorithm for an active headrest. The performance of the proposed method has been evaluated experimentally using periodic and band limited white noise. Improved noise control performance has been observed for both periodic and broadband noise. The effect of causality constraint of the performance of the algorithm has also been tested.

Active control of impulsive scattered sound based on virtual sensing

Active control method is an important way to reduce the scattered radiation of 3D objects and render them acoustically invisible. Traditional active control requires the error sensor to be sufficiently close to the target area because the zone of quiet (ZoQ) is generated around the sensor; thus, the scattered noise cannot be eliminated at the remote position in situations wherein the microphone can only be placed near the scatterer. To solve this dilemma, this study proposed two methods of predicting and controlling the scattered sound, combined with adaptations of two commonly used virtual sensing (VS) methods: the auxiliary-filter-based virtual-sensing (AF-VS) and the remote microphone technique (RMT). Both methods are based on the correlation between signals to predict the scattered noise; therefore, ZoQ can be transferred to the non-near-field target position with fewer physical microphones. Further, the causal constraints need not be satisfied in the former method, which uses pre-trained auxiliary filters to implicitly estimate and control the scattered pressure at the target point. Whereas, the latter explicitly estimates the scattered component instead of the total sound pressure with the pre-modeled observation matrix, thereby avoiding the causality limitation of the RMT and reducing one estimation process. Experiments were conducted to suppress impulsive scattered sound using a dual-channel feedforward adaptive ANC system, and the attenuation performances of the two proposed methods were compared with those of the direct control of the desired noise with the optimal Wiener solution and the method without combining the VS. The results demonstrated the feasibility and effectiveness of the two active control methods for scattered sound using virtual sensors. Moreover, in the impulsive scattered sound control with a 300 Hz-1000 Hz carrier, the average noise reduction (NR) at the observation points reached approximately 18 dB, and a more stable and excellent performance was obtained using the latter method based on RMT. Therefore, this research provides a solution for the case of limited error sensor placement in scattered sound active control systems; and in addition, can be used to guide the choice of VS methods in scattered sound control applications.

A Study on Improving the Robustness of Virtual Sensing Methods in ANC Systems

Active noise control (ANC) system generates anti-noise with the same amplitude and opposite phase to reduce unwanted acoustic noise based on acoustic wave superposition. In general, ANC can generate a zone of quiet (ZoQ) around the error microphone position. However, if the microphone cannot be placed at the desired position we want to realize noise reduction, the reduction effect may not be sufficient in the desired position. In such a case, we need to use the virtual sensing method to move the ZoQ to the desired position. Several virtual sensing methods have been proposed, each with different robust performance due to the different way the system is designed. It is known that the noise reduction is degraded when there are perturbations in the environment such as plant and noise. In this paper, we investigate how to improve the robust performance of the virtual sensing method.

Multichannel two-gradient direction filtered reference least mean square algorithm for output-constrained multichannel active noise control

Multichannel active noise control is a proven and long-standing technique for achieving a large zone of quiet in enclosures and open spaces. In practice, the multichannel active noise control system encounters more significant output saturation nonlinearity than the single-channel system as a result of the increased number of multiple output amplifiers, which not only degrades noise reduction performance but also compromises system stability. Conventional output-saturation solutions are mainly designed for single-channel applications, and their high computational complexity prevents them from being extended to multichannel systems. Hence, this paper proposes a novel two-gradient direction multichannel filtered reference least mean square (FxLMS) method that effectively avoids the saturation problem by restricting the output to the desired range and marginally increasing the computation over the conventional Multichannel FxLMS algorithm. In addition, this paper investigates the step sizes of the algorithm and proposes a computation-efficient variable step-size strategy to reduce further the steady-state error caused by the varying gradient directions. Finally, the efficacy of the proposed algorithm is demonstrated through numerical simulations using measured primary and secondary paths.

An online identification method for virtual sensing in ANC system

Active noise control (ANC) system generates anti-noise with the same amplitude and opposite phase to reduce unwanted acoustic noise based on acoustic wave superposition. In general, ANC can generate a zone of quiet (ZoQ) around the error microphone position. However, if the microphone cannot be placed at the desired position, the reduction effect may not be sufficient in the desired position. In such a case, we need to use the virtual sensing method to achieve noise reduction at the desired position without placing the physically microphone. In virtual sensing method, secondary path variation can significantly degrade performance, and secondary path modeling errors can cause the system to diverge. In this presentation, we introduce a new online identification method for auxiliary filter-based virtual sensing in ANC system. Virtual sensing with online identification requires re-estimation of not only the secondary path but also the auxiliary filters. The proposed online identification method has the capability of online identification of auxiliary filters in addition to the conventional online identification of secondary paths. We show that through computer simulations the noise reduction can be maintained by applying the proposed online identification method even when the secondary path varies.

Active headrests with selective delayless subband adaptive filters in an aircraft cabin

Active noise control (ANC) systems try to attenuate noise by generating local zones of quiet. In this paper local ANC systems in a relevant environment are investigated, usingry conditions derived from environments like cars, planes or trains. The experimental setup consists of a wooden aircraft cabin mock-up and an active headrest system using two actuators (loudspeakers) and two sensors (microphones) placed in a headrest to create individual local zones of quiet at the ears of the occupant.Various concepts for active headrest systems are taken into account, such as multichannel controller concepts, virtual sensing techniques and delayless subband adaptive filters, focusing on the delayless subband adaptive filters. Based on these studies a new active headrest system is proposed, that is capable of generating a 10 dB zone of quiet for different non-stationary broadband disturbances, while compensating head movement. Using a selective delayless subband adaptive filter and off the shelf microphones and loudspeakers, the overall hardware demand is kept to a minimum allowing for a low cost application.

An active noise control casing using the multi-channel feedforward control system and the relative path based virtual sensing method

Active noise control (ANC) is a noise reduction technique based on acoustic wave superposition. The sound pressure level is reduced in a zone of quiet (ZoQ) by an anti-noise wave transmitted from the control source. The anti-noise wave has the same amplitude and opposed phase of the noise wave. An error microphone is conventionally placed at the target ZoQ to monitor the sound pressure level, forming a closed-loop control in an ANC system. However, due to application constraints or physical limitations, the error microphone sometimes cannot be placed at the target ZoQ. Virtual sensing (VS) methods are developed for such situations. There are two most commonly used VS methods. They are the auxiliary filter based VS (AF-VS) method and the remote microphone based VS (RM-VS) method. The AF-VS method preserves the information regarding the optimal control filter that can achieve the maximum noise reduction at the target ZoQ. The RM-VS method estimates the disturbance signal at the target ZoQ based on remote measurements. In this paper, we propose a new VS method, the relative path based VS (RP-VS) method, which estimates both the disturbance signal and the anti-noise signal at the target ZoQ. A theoretical analysis is provided to demonstrate that under different assumptions of varying acoustic paths, the RP-VS method can behave in the same way as the AF-VS method or the RM-VS method. Simulation results validate this theoretical analysis and demonstrate that improved noise reduction can be achieved by the RP-VS method when the noise frequency varies. Lastly, an ANC casing is built up with the RP-VS method to reduce a varying broadband fan noise. The RP-VS method is validated to be as effective as the AF-VS method and the RM-VS method with the implementation of the ANC casing.

On FxLMS Scheme for Active Noise Control at Remote Location

The conventional adaptive noise control algorithms create a zone of quiet at the location of an error microphone. Several virtual active noise control (ANC) algorithms have been suggested in the literature to introduce a more flexible positioning of the zone of quiet. The main objective of the proposed ANC system in our research is to minimize the noise disturbance on human hearing by considering the auditory system characteristics. We have introduced the human perception of acoustic disturbances in our ANC modeling. This paper presents a novel approach based on a psychoacoustically enabled remote ANC algorithm to minimize the acoustic noise at a zone away from the error microphone location. This approach depends on modeling the transfer function between the error microphone and the remote zone of quiet. The developed model has been implemented using an FPGA real-time module, and the performance is validated at four different remote locations. Noise-weighting filters are incorporated to observe the psychoacoustic characteristics of the proposed model. In the experimental setup, the acoustic noise at the physical and remote locations are monitored to quantify the noise reduction characteristics of the proposed algorithm. The simulation and experimental results show a noise reduction of 15-18 dBs has been achieved, which is an average improvement of 5-8 dBs over the standard ANC model.

Analysis of the complex sound power in the near field of spherical loudspeaker arrays

Over the past decade, compact spherical loudspeaker arrays have emerged as directivity controlled electroacoustic sources. The usual approach to control such a device is to provide it with a finite set of pre-programmed directivities corresponding to spherical harmonics or to the acoustic radiation modes (ARMs) of the array. However, these orthogonal bases are not suited for the acoustic control in the near field, which requires a number of spherical harmonics larger than the number of loudspeakers, and modes that take into account the reactive part of the complex sound power, not only the active part as is the case for the conventional ARMs. This paper presents a theoretical and numerical study of alternative ARM approaches applied to spherical loudspeaker arrays that consider both components of the sound power, with focus on the arrangements following the regularity of the five Platonic solids. For such arrays, it is shown that the active, reactive, and the so-called generalized ARMs form exactly the same basis, which does not depend either on frequency, source-receptor distance or on the vibration pattern and shape of the individual loudspeakers, provided that these are identical and axisymmetric. Closed-form expressions for the modal active and reactive efficiencies are derived and evaluated for the Platonic solid loudspeakers as a function of frequency and source-receptor distance. In addition, an application example in local active noise control is presented, in which a spherical loudspeaker array is used as a secondary source aiming at producing a zone of quiet in its near field.

Head tracking extends local active control of broadband sound to higher frequencies

Local active sound control systems provide useful reductions in noise within a zone of quiet which only extends to about one tenth of an acoustic wavelength. If active control is required above a few hundred hertz, this generally limits the movement of a listener to unrealistically small changes in head position. We describe a local active sound control system using a fixed array of monitoring microphones, in which the pressures at the ear positions are estimated from these microphone signals using head position information from an optical head tracker. These signals are then actively controlled to give robust attenuation at the ear positions, even as the listener moves their head. Feedforward control provides selective attenuation of noise and broadband attenuation of around 20 dB is measured up to excitation frequencies of 1 kHz under favourable conditions, with head tracking achieved in a few seconds. The active control performance is thus comparable with that achieved with active headphones, but without the listener having anything attached to their head.

다채널 실시간 능동 소음제어 시스템을 이용한 정숙공간 성능개선

3차원 잡음환경에서 정숙공간은 현실적으로 매우 중요한 문제이다. 본 논문은 3차원 잡음 환경에서 정숙 공간의 성능을 크기와 소음제거 면에서 향상시키는 다채널 실시간 능동소음 제어시스템의 개발과 구현을 다루고 있다. 제안된 능동소음제어 시스템은 delay-compensated Filtered-X Least Mean Square (FXLMS) 알고리즘을 적용한다. 이와 같은 시스템의 실시간 적용을 위해서 TMS320C6713 DSP 프로세서 기반으로 설계되었다. 제안된 실시간 다채널 능동소음제어기의 성능평가는 100 ~ 500 Hz 범위의 다양한 잡음 환경에서 잡음제거를 수행하고, 정숙공간에서 음압레벨(Sound Pressure Level, SPL)측정하여 평가하였다. 실험결과는 정숙공간의 크기는 만족스러우며 최대 24 dB의 소음 감쇄가 성공적으로 생성된 것을 보여준다. Generation of a quiet zone in noisy environment is undoubtedly of considerable realistic significance. This paper describes development and implementation of a multichannel real-time active noise control (ANC) system for 3 dimensional noisy environment to enhance the quiet zone performance in terms of size and noise cancellation gain. The proposed ANC system employes a multichannel delay-compensated filtered-X least mean square (FXLMS) algorithm; its real-time implementation is designed in TMS320C6713 digital signal processor (DSP) board. The system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz, and the performance is then illustrated by measuring the quiet zone in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone of quiet with satisfactory size and maximum 24 dB noise attenuation is successfully generated.

Characteristics of Relocated Quiet Zones Using Virtual Microphone Algorithm in an Active Headrest System

This study displays theoretical and experimental investigation on the characteristics of the relocated zone of quiet by a virtual microphone (VM) based filtered-x LMS (FxLMS) algorithm which can be embedded in a real-time digital controller for an active headrest system. The attenuation changes at the relocated zones of quiet by the variation of the distance between the ear and the error microphone are mainly examined. An active headrest system was implemented for the control experiment at a chair and consists of two (left and right) secondary loudspeakers, two error microphones, two observer microphones at ear positions in a HATS, and other electronics including a dSPACE 1401 controller. The VM based FxLMS algorithm achieved an attenuation of about 22 dB in the control experiment against a narrowband primary noise by the variation of the distance between the ear and the error microphone. The important factors for the algorithm are discussed as well.

좌석의 정음공간 형성을 위한 가상마이크로폰 기반 능동음향제어 기법 연구

ABSTRACT In this study, theoretical and experimental analyses were performed for creating and moving the zone of quiet(ZoQ) to the ear location of a sitter by using active sound control technique. As the ZoQ is actively created at the location of the error microphone basically with an active sound con-trol system using an algorithm such as the filtered-x least mean square(FxLMS), the virtual micro-phone control(VMC) method was considered to move the location of the ZoQ to around the sitter`s ear. A chair system with microphones and loudspeakers on both sides was manufactured for the ex-periment and thus an active headrest against the swept narrowband noise as the primary noise was implemented with a real-time controller in which the VMC algorithm was embedded. After the con-trol experiment with and without the VMC method, the location variation of the ZoQ by analyzing the error signals measured by the error and the virtual microphones. Therefore, it is observed that the FxLMS with the VMC technique can provide the re-location of the ZoQ from the error micro-phone location to the virtual microphone location. Also it is found that the amount of the attenu-ation difference between the two locations was small.

Modeling local active sound control with remote sensors in spatially random pressure fields.

A general formulation is presented for the optimum controller in an active system for local sound control in a spatially random primary field. The sound field in a control region is selectively attenuated using secondary sources, driven by reference sensors, all of which are potentially remote from this control region. It is shown that the optimal controller is formed of the combination of a least-squares estimation of the primary source signals from the reference signals, and a least-squares controller driven by the primary source signals themselves. The optimum controller is also calculated using the remote microphone technique, in both the frequency and the time domains. The sound field under control is assumed to be stationary and generated by an array of primary sources, whose source strengths are specified using a spectral density matrix. This can easily be used to synthesize a diffuse primary field, if the primary sources are uncorrelated and far from the control region, but can also generate primary fields dominated by contributions from a particular direction, for example, which is shown to significantly affect the shape of the resulting zone of quiet.

Quiet Zone Design in Diffuse Fields Using Ultrasonic Transducers

This paper presents quiet zone design using ultrasonic transducers for local active control in pure tone diffuse fields. Most of researches in local active noise control used conventional loudspeakers for the secondary sources to produce quiet zones. Recently ultrasonic transducers have been used for the secondary sources to control the plane wave in active noise control. However there is no research related to active noise control in diffuse fields using ultrasonic transducers. Therefore this study uses ultrasonic transducers for the secondary sources to control the diffuse fields. The quiet zone produced using ultrasonic transducers in single tone diffuse fields has been analyzed through simulations in this work. The results showed that quiet zones created using ultrasonic transducers were larger than those created using conventional loudspeakers. This is due to the fact that the audible sound pressure produced by the ultrasonic transducers decays slowly with the distance. Therefore the secondary field created by an ultrasonic transducer could fit the primary field better and the larger zone of quiet could be obtained using the ultrasonic transducer. Also the audible sound produced by the ultrasonic transducers is directional; therefore the sound pressure amplification outside the quiet zones was lower.

Adaptive signal processing algorithms for creating spatial zones of quiet

Available adaptive active noise control (ANC) algorithms can only minimize the noise level at a point that an error microphone is placed. Consequently, a zone of quiet around this microphone is produced as a byproduct. However, they cannot technically control or, even, monitor the noise level within the zone of quiet unless they use several sensors. They cannot also control the shape and the extension of the quiet zone by using only the error microphone. This paper develops a signal processing framework for the derivation of adaptive ANC algorithms that can directly create a controllable zone of quiet in monochromatic noise fields using a single error microphone. It is shown that by adding a filter to the standard ANC structure, a controllable zone of quiet is created. The transfer function of this filter is obtained using an accurate mathematical analysis. It is also shown that the extension of the zone of quiet can be controlled by tuning this filter. The implementation of the proposed system requires no additional hardware, rather than those required for traditional ANC systems. The validity of the results are discussed by using numerical analysis. Also, the performance of the proposed system is practically verified.

ESPRESSO: The next European exoplanet hunter

AbstractThe acronym ESPRESSO stems for Echelle SPectrograph for Rocky Exoplanets and Stable Spectroscopic Observations; this instrument will be the next VLT high resolution spectrograph. The spectrograph will be installed at the Combined‐Coudé Laboratory of the VLT and linked to the four 8.2 m Unit Telescopes (UT) through four optical Coudé trains. ESPRESSO will combine efficiency and extreme spectroscopic precision. ESPRESSO is foreseen to achieve a gain of two magnitudes with respect to its predecessor HARPS, and to improve the instrumental radial‐velocity precision to reach the 10 cm s–1 level. It can be operated either with a single UT or with up to four UTs, enabling an additional gain in the latter mode. The incoherent combination of four telescopes and the extreme precision requirements called for many innovative design solutions while ensuring the technical heritage of the successful HARPS experience. ESPRESSO will allow to explore new frontiers in most domains of astro‐physics that require precision and sensitivity. The main scientific drivers are the search and characterization of rocky exoplanets in the habitable zone of quiet, nearby G to M‐dwarfs and the analysis of the variability of fundamental physical constants. The project passed the final design review in May 2013 and entered the manufacturing phase. ESPRESSO will be installed at the Paranal Observatory in 2016 and its operation is planned to start by the end of the same year. (© 2014 WILEY‐VCH Verlag GmbH & Co. KGaA, Weinheim)

Local and global active noise control using a parametric array loudspeaker

This paper deals with local as well as global active noise control (ANC) using a parametric array loudspeaker (PAL) possessing intriguing properties: sharp directivity, low sound pressure decay by distance, capability of steering directivity, etc. After briefing some properties of a PAL necessary for ANC, this paper presents pinpoint control using a PAL for suppressing the sound pressure at a designated location, hence local control. It is shown that unlike conventional ANC in which a voice coil loudspeaker is used, the pinpoint control may achieve the sound pressure suppression without causing spillover. Using the same sound control source, PAL, this paper then refers to global control, termed trivial control in the art, enabling one to generate a global zone of quiet. The trivial control strategy falls into a category of acoustic power control based on a trivial condition requiring the collocation of a primary source and a control source. The trivial condition formerly avoided because of literally trivial may be implemented due to the property of a PAL, thereby enhancing the control effect, theoretically infinity. Two kinds of control strategy are then demonstrated with a view to fulfilling the trivial condition for global control.

A Novel Method of Designing Quiet Zones in Diffuse Fields

This paper presents the analysis of quiet zones in broadband diffuse fields using two- and three-channel systems. An optimization method combining a neural network with a search algorithm has been used to design quiet zones in broadband diffuse fields. Simulation results using the proposed method are presented. Experiments are also carried out to validate the simulation results. The results show that a good attenuation in the desired zone of quiet over space and frequency is achieved using a two-channel system. However, a better performance could be achieved using a three-channel system. Also, the shape and location of quiet zones over space and frequency could be controlled by using the proposed method. The results also show that the shape and size of 10-dB quiet zones are both similar in simulations and experiments. The main contributions of the paper are that a novel method of designing quiet zones in broadband diffuse fields is proposed and the performance of the active systems is analyzed through computer simulations and experiments.

629 一次音源が移動する音場での仮想マイク法による能動音響制御

Local active noise control systems create a zone of quiet around a physical error sensor location. The zone of quiet is generally small, and as such, the physical error sensor is placed at the desired location of sound attenuation, which is usually inconvenient. Virtual sensing algorithms have been developed to aim for creating the zone of quiet at virtual sensor location that is remote from the physical error sensor. However the sound attenuation of the traditional virtual sensing algorithms worsen by change of primary source location that alter the transfer characteristics between a virtual sensor and a physical sensor. This paper presents a method to approximate the transfer characteristics between a virtual sensor and a physical sensor by superposition of several transfer functions calculated from acoustic paths using positional information of primary source. The transfer function is expressed a dead-time component and a coefficient vector for amplitude correction calculated by Maekawa's curve. The effect of control using the approximated transfer function in numerical simulation indicates that around 5dB of attenuation can be achieved.