Strategies for bass enhancement in Multiactuator Panels for Wave Field Synthesis

In this paper, an extension of the useful bandwidth in the low frequency band is applied to compensate the inherent poor bass response of Multiactuator Panels. These are a special type of flat panel loudspeakers that are commonly used to reproduce spatial audio under the Wave Field Synthesis system. The proposed algorithm combines two strategies: first, a dynamic electrical equalization, applied to additional exciters, which are carefully positioned to excite the lower frequency modes of the panel; second, a psycho-acoustical approach, taking profit of the behavior of human hearing based on the missing fundamental principle. For comparison purposes, the shelving and peak equalizations are also applied to the MAP prototypes developed in this paper. Objective and subjective results show that the combined approach results in an effective extension in the low frequency end of MAP with very low levels of distortion, outperforming conventional equalization methods, with better low frequency behavior, and hence better audio quality perception.     

Optimization of orthotropic distributed-mode loudspeaker using attached masses and multi-exciters

Based on the orthotropic model of the plate, the method to optimize the sound response of the distributed-mode loudspeaker (DML) using the attached masses and the multi-exciters has been investigated. The attached masses method will rebuild the modes distribution of the plate, based on which multi-exciter method will smooth the sound response. The results indicate that the method can be used to optimize the sound response of the DML.     

Experimental implementation of a low-frequency global sound equalization method based on free field propagation

An experimental implementation of a global sound equalization method in a rectangular room using active control is described in this paper. The main purpose of the work has been to provide experimental evidence that sound can be equalized in a continuous three-dimensional region, the listening zone, which occupies a considerable part of the complete volume of the room. The equalization method, based on the simulation of a progressive plane wave, was implemented in a room with inner dimensions of 2.70 m × 2.74 m × 2.40 m. With this method, the sound was reproduced by a matrix of 4 × 5 loudspeakers in one of the walls. After traveling through the room, the sound wave was absorbed on the opposite wall, which had a similar arrangement of loudspeakers, by means of active control. A set of 40 digital FIR filters was used to modify the original input signal before it was fed to the loudspeakers, one filter for each transducer. The optimal arrangement of the loudspeakers and the maximum frequency that can be equalized is analyzed theoretically in this paper. The presented experimental results show that sound equalization was possible from 10 Hz to approximately 425 Hz in the listening zone. A flat frequency response with deviations within ±5 decibels from the desired value was achieved. A higher demanding performance with deviations within ±1.5 decibels from a flat frequency response was attained in the interval between 20 Hz and 280 Hz. At the same time, the impulse response was quite well approximated to a delayed delta function in the listening zone. Examples of the spatial distribution of the sound field are also shown.     

A synthesis approach for reproducing the response of aircraft panels to a turbulent boundary layer excitation

Random wall-pressure fluctuations due to the turbulent boundary layer (TBL) are a feature of the air flow over an aircraft fuselage under cruise conditions, creating undesirable effects such as cabin noise annoyance. In order to test potential solutions to reduce the TBL-induced noise, a cost-efficient alternative to in-flight or wind-tunnel measurements involves the laboratory simulation of the response of aircraft sidewalls to high-speed subsonic TBL excitation. Previously published work has shown that TBL simulation using a near-field array of loudspeakers is only feasible in the low frequency range due to the rapid decay of the spanwise correlation length with frequency. This paper demonstrates through theoretical criteria how the wavenumber filtering capabilities of the radiating panel reduces the number of sources required, thus dramatically enlarging the frequency range over which the response of the TBL-excited panel is accurately reproduced. Experimental synthesis of the panel response to high-speed TBL excitation is found to be feasible over the hydrodynamic coincidence frequency range using a reduced set of near-field loudspeakers driven by optimal signals. Effective methodologies are proposed for an accurate reproduction of the TBL-induced sound power radiated by the panel into a free-field and when coupled to a cavity.     

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.     

Acoustical performance of an electrostrictive polymer film loudspeaker

A new type of loudspeaker that generates sound by means of the electrostrictive response of a thin polymer film is described. Electrostrictive polymer film (EPF) loudspeakers are constructed with inexpensive, lightweight materials and have a very low profile. The films are typically silicone and are coated with compliant electrodes to allow large film deformations. Acoustical frequency response measurements from 5 x 5 cm (planar dimensions) prototype EPF loudspeakers are presented. Measurements of harmonic distortion are also shown, along with results demonstrating reduced harmonic distortion achieved with square-root wave shaping. Applications of EPF loudspeakers include active noise control and general-purpose flat-panel loudspeakers.     

Spatially extended sound equalization in rectangular rooms.

The results of a theoretical study on global sound equalization in rectangular rooms at low frequencies are presented. The zone where sound equalization can be obtained is a continuous three-dimensional region that occupies almost the complete volume of the room. It is proved that the equalization of broadband signals can be achieved by the simulation of a traveling plane wave using FIR filters. The optimal solution has been calculated following the traditional least-squares approximation, where a modeling delay has been applied to minimize reverberation. An advantage of the method is that the sound field can be estimated with sensors placed in the limits of the equalization zone. As a consequence, a free space for the listeners can be obtained.     

Flat-response sound source technique for using the two-microphone method in an impedance tube

In this letter, to improve the acoustic impedance measurement in a tube using the two-microphone method, a sound source technique is developed that is capable of generating incident sound waves of uniform intensity over the entire frequency range of interest. The basic principle is to use a digitally synthesized compensation input signal to achieve a flat response of the sound source provided by one or more loudspeakers. The present sound source technique has been demonstrated to be very effective and easily implemented with the current impedance tube testing systems.     

Enhancing low frequency sound transmission measurements using a synthesis method

The characterization of low frequency sound transmission between two rooms via a flexible panel is investigated experimentally in this work. Previously, the individual effects of the transmission suite on the measured sound reduction index have been studied analytically, and the results have been compared with the ideal case of having free field radiation conditions on both sides of the panel. A new approach is proposed using a near-field array of loudspeakers driven by a set of optimized signals such that a diffuse pressure field is reproduced on the surface of the partition to be tested. The practical effectiveness of this method is assessed when using a set of 16 acoustic sources located in the source reverberant room in close proximity to an aluminium panel. The experimental results obtained confirm the dependence of the characterized sound reduction index on the particular test chamber considered in the low frequency range. They also validate the proposed synthesis method for providing an estimate that only depends on the properties of the partition itself.     

内部激励源二自由度结构扬声器振动特性分析

有别于多数基于简单振子结构的扬声器,针对一类内部激励源二自由度(IE2DOF)结构的扬声器,用类比线路图法建立集总参数模型,计算分析了这种结构的频率响应,同时计算了其固有共振频率和固有反共振频率。使用叠加法分析内部激励源对频响的影响。最后实际测量和理论计算吻合,进一步支持了理论模型,揭示了IE2DOF结构扬声器的振动特性。     

A multichannel and multiple position adaptive room response equalizer in warped domain: Real-time implementation and performance evaluation

Room response equalization systems are used for improving the listening experience in cinema theatres, home theatres, car hi-fi systems. In this paper, an adaptive multichannel and multiple position room response equalization system and its real-time implementation are described. An adaptive and accurate estimation of the room responses is provided introducing a normalized least mean square optimization approach with a variable step-size, and taking advantage of an interchannel coherence reduction technique based on the missing fundamental phenomenon. Then, the equalizer is designed in warp frequency domain for improving equalization in the low frequency region, reducing the computational cost of the design procedure, and deriving an algorithm capable of working in real time. Indeed, a real-time implementation of the proposed adaptive equalizer has been obtained on NU-Tech framework and has been used in order to provide a deep objective and subjective evaluation of the equalization system. The results of these evaluations illustrate the effectiveness of the proposed approach, also in comparison with other techniques of the state of the art.     

IS A SIMPLE SUPPORT REALLY THAT SIMPLE?

A very frequently applied boundary condition in problems involving modelling the dynamic behaviour of beams and panels is a simple support. This type of boundary condition is particularly used in the research area of active control of vibrations on panels. However, the practical implementation of this boundary condition in the laboratory is not at all trivial, especially when high accuracy is required.In this paper, various possible practical implementations of a simply supported panel are considered. A suitable method is selected and investigated experimentally. The results from a forced vibration test are compared with those obtained from theory in order to verify the relative accuracy of the approach.The effect of the relative distance between the panel surface and a solid boundary on the fundamental resonant response frequency is experimentally investigated, as well as changes produced by fluctuations in the laboratory temperature.     

Least squares approach in wavenumber domain for sound field recording and reproduction using multiple parallel linear arrays

A novel signal processing method is proposed for sound field recording and reproduction using multiple parallel linear microphone and loudspeaker arrays. In sound field recording and reproduction, the problem is how to calculate the transfer filters that transform the signals recorded by microphones into the driving signals of the loudspeakers. The proposed method is based on the spatial Fourier transform in the horizontal angle combined with the least squares (LS) approach in the elevation angle. In the proposed method, the signals recorded by each linear microphone array and those that drive each loudspeaker array are decomposed into the wavenumber domain by the spatial Fourier transform in the horizontal direction. The transfer filters are then calculated by the LS approach in the wavenumber domain. As a result, the size of the matrix of each transfer function in the wavenumber domain is much smaller than that of the conventional LS approach in the temporal frequency domain (LSTF), and well-conditioned stable transfer filters can be obtained with low computational cost without regularization. Computer simulation results show that the proposed method reconstructed a sound field around the control points as accurately as the conventional LSTF.     

参量阵扬声器在管道噪声控制中的研究*

为了解决管道有源噪声控制中声反馈造成的系统复杂度和计算量的增加,文中引入参量阵扬声器作为次级声源,利用其强指向性减小控制系统的声反馈。为了验证该方法可行性,本文分别在直管和L管中,对600 Hz单频噪声和频率范围为500 Hz~1000 Hz的窄带噪声进行了管道有源噪声控制,同时测量了参量阵扬声器的管内声场和降噪范围。结果表明,参量阵扬声器声反馈小,在没有声反馈补偿的条件下对单频噪声的降噪效果基本达到了声反馈补偿条件下普通扬声器的降噪效果,对窄带噪声的降噪效果稍差。此外,通过测量管道声场和降噪量,确定了参量阵扬声器的降噪区域为误差传感器下游整个管道,降噪面积为管道整个截面。这说明参量阵扬声器作为次级声源降低了系统的复杂度和算法的计算量,并取得了较好的降噪效果。     

基于少模光纤的模分复用系统多输入多输出均衡与解调

针对光纤模分复用传输系统中模式耦合串扰问题, 设计并制备了一种新型少模光纤, 其较高的模式差分群延时保证各模式信道独立传输.在此基础上, 提出一种级联多输入多输出(MIMO)延时均衡算法, 进一步减少源于模式复用器和解复用器的模式串扰, 提高基于少模光纤的模分复用传输系统的传输距离和传输容量.与传统MIMO均衡算法相比, 级联MIMO延时均衡算法在没有显著增加计算复杂度的条件下, 能够应用于模式差分群延时很大的模分复用传输系统.对单信道传输速率为 40 Gbps的四相相移键控两模复用传输系统进行仿真, 经40 km少模光纤传输后, 采用级联MIMO均衡算法较普通MIMO均衡算法有1.7 dB的质量因子的提升. 仿真结果证明, 使用少模光纤和级联MIMO延时均衡算法能够有效地消除模分复用信号间的串扰, 有望在下一代大容量光纤传输系统中获得 推广应用. 关键词： 模分复用 少模光纤 模式差分群延时 多输入多输出均衡     

气流扬声器振动系统的设计原理

本文用等效电路方法分析了电动气流扬声器振动系统的特性,求得了气流扬声器的输入阻抗和位移振幅的频率特性,并把结果用于典型扬声器,说明用等效电路分析指导气流扬声器设计的作用。气流扬声器振动系统所需的电功率与其机械共振频率的四次方成正比,其声辐射主要限于共振频率以下,因而高频特性较差。求得了在共振频率以下,频率响应平直的最佳条件,实验表明,在最佳条件下,气流扬声器的高频特性随着磁隙中的磁感应强度的增加仍稍有提高。实验结果符合理论分析,测得气流扬声器喇叭喉部的声压级在不同情况下达181分贝或184.8分贝(零分贝等于2×10^-5牛顿/米²)。     

基于微米颗粒填充的粉末低通滤波器仿真

粉末滤波器是一种可应用于超导量子计算等低温实验系统的低通滤波器, 随着频率的升高滤波器会对信号产生急剧的有效衰减, 粉末滤波器的填充材料、 中心导线长度、 线径和结构等参数会对它的频率响应特性产生影响. 本文利用 CST 微波工作室基于微米颗粒填充的方式, 首次对不同中心导线结构的粉末滤波器性能进行三维仿真研究, 仿真结果与实测结果对比后的定性一致性, 说明仿真方法的可行性和准确性. 仿真方法的提出可以有效地加快滤波器结构的迭代速度, 从而降低研发周期和成本, 方便可应用于超导量子计算的新一代粉末滤波器的开发.     

On the acoustic radiation modes of compact regular polyhedral arrays of independent loudspeakers

Compact spherical loudspeaker arrays can be used to provide control over their directivity pattern. Usually, this is made by adjusting the gains of preprogrammed spatial filters corresponding to a finite set of spherical harmonics, or to the acoustic radiation modes of the loudspeaker array. Unlike the former, the latter are closely related to the radiation efficiency of the source and span the subspace of the directivities it can produce. However, the radiation modes depend on frequency for arbitrary distributions of transducers on the sphere, which yields complex directivity filters. This work focuses on the most common loudspeaker array configurations, those following the regular shape of the Platonic solids. It is shown that the radiation modes of these sources are frequency independent, and simple algebraic expressions are derived for their radiation efficiencies. In addition, since such modes are vibration patterns driven by electrical signals, the transduction mechanism of compact multichannel sources is also investigated, which is an important issue, especially if the transducers interact inside a shared cabinet. For Platonic solid loudspeakers, it is shown that the common enclosure does not lead to directivity filters that depend on frequency.     

The time domain response of some systems for sound reproduction

The perception in a listener of the existence of a “virtual” source of sound at a prescribed spatial position can be produced by ensuring that the acoustic signals at the listener's ears faithfully replicate those that would be produced by a “real” source at the same position. When loudspeakers are used to transmit the signals, it is necessary to pass the signals intended for presentation at the listener's ears through a matrix of filters that provide the inverse of the matrix of transfer functions that relates the loudspeaker input signals to the listener's ear signals. The characteristics of such filter matrices are profoundly influenced by the conditioning of the matrix to be inverted. This filter design problem is reviewed here by representing the loudspeakers as simple point monopole sources the head of the listener as a rigid sphere. The case of a virtual acoustic imaging system that uses two loudspeakers in order to reproduce the signals at the two ears is first described in some detail and previous work is reviewed. It is confirmed that the time domain response of the reproduced field is of long duration at frequencies where the inversion problem is ill-conditioned. The influence of the presence of the listener's head on this time domain behaviour is also evaluated. The principle is then extended to four input-four output reproduction systems and the computational model is used to explain some previous experimental observations. Finally, the conditioning of five input-four output systems is also examined and shown to have some potentially desirable characteristics.     

Efficient processing of acoustic signals for high-rate information transmission over sparse underwater channels

For underwater acoustic channels where multipath spread is measured in tens of symbol intervals at high transmission rates, multichannel equalization required for bandwidth-efficient communications may become prohibitively complex for real-time implementation. To reduce computational complexity of signal processing and improve performance of data detection, receiver structures that are matched to the physical channel characteristics are investigated. A decision-feedback equalizer is designed which relies on an adaptive channel estimator to compute its parameters. The channel estimate is reduced in size by selecting only the significant components, whose delay span is often much shorter than the multipath spread of the channel. Optimal coefficient selection (sparsing) is performed by truncation in magnitude. This estimate is used to cancel the post-cursor ISI prior to linear equalization. Spatial diversity gain is achieved by a reduced-complexity pre-combining method which eliminates the need for a separate channel estimator/equalizer for each array element. The advantages of this approach are reduction in the number of receiver parameters, optimal implementation of sparse feedback, and efficient parallel implementation of adaptive algorithms for the pre-combiner, the fractionally-spaced channel estimators and the short feedforward equalizer filters. Receiver algorithm is applied to real data transmitted at 10 kbps over 3 km in shallow water, showing excellent results.