An algorithm for designing a broadband active sound control system

Hitherto the filtering techniques used in active sound control systems have been analogue: these are rapidly becoming obsolete because of their lack of stability and versatility. The microprocessor gives digital techniques an edge both on grounds of convenience and economics; moreover the performance of digital filtering is far superior to that of analogue methods. An algorithm is presented which can be used to assess the optimum filter characteristics required for active sound control systems with a single degree of freedom. In the process the system is subjected to three random noise tests which directly yield the characteristics of the filter. The algorithm has been tried out in practical applications and shown to be both quick and convenient to use.     

A demonstration of active control of broadband sound

This paper describes a short experiment undertaken to demonstrate how easily broad-band active control of sound can be used to tackle real industrial problems. Ten decibles of the low frequency sound entering an anechoic chamber through a lobby was blocked by using a single degree of freedom system. The experiment was set up, tested and demonstrated in a day.     

Development of adaptive algorithm for active sound quality control

This paper develops an active sound-quality control (ASQC) system based on the active noise equalization (ANE) technique, and optimizes it with the filtered-error least mean square (FELMS) algorithm and normalized reference signal generator. The ASQC system controls the sound quality of products such as engines by changing the amplitudes of harmonics. This optimized system uses the FELMS algorithm to limit disturbances in the passband caused by uncorrelated interferences with high gains in the secondary path, thereby increasing the system stability. It achieves fast convergence by normalizing the amplitudes of internally generated sinusoids in the reference signal according to the magnitude response of the secondary path at the corresponding frequencies. Computer simulations demonstrate the desired spectral shaping capability with faster convergence and reduced passband disturbance.     

采用听觉滤波器的宽带MUSIC声源定位方法

在分析了采用短时傅里叶变换的宽带MUSIC声源定位算法(SF-MUSIC)存在问题的基础上,提出了一种采用听觉滤波器的宽带MUSIC声源定位算法(AF-MUSIC)。该算法使用听觉滤波器组对传声器阵列接收到的信号进行不等带宽分解后,在各个频率通道上使用MUSIC算法进行声源定位,并结合子区间频数估计法得出最终定位结果。对算法进行的实验评估表明,在不同声源类型条件下,相比SF-MUSIC算法,AF-MUSIC算法的平均估计误差减少2.5479°,有效地提高了声源波达方向估计的精度。     

Auditory filter based broadband MUSIC algorithm for sound source localization

Based on the analysis of the shortcomings of broadband MUSIC algorithm with short-time Fourier transform （SF-MUSIC） for sound source localization, a broadband MUSIC algorithm with auditory filter （AF-MUSIC） was proposed. The proposed algorithm first em- ploys auditory filter bank to decompose the signals received on the microphone array, and then locates the sound source with MUSIC algorithm over every frequency channel. At last, by combining with the subinterval frequency estimation, the final localization result is gained. Evaluations on the proposed algorithm prove that comparing with the SF-MUSIC algorithm, the AF-MUSIC algorithm decreases the average error of the estimation results with 2.5479 de- gree in different source conditions. The accuracy of sound source DOA estimation is enhanced effectively.     

An adaptive controller for multivariable active noise control

The control of vibration through the mounts of rotating machines can be achieved by actively generating cancelling forces from shakers located close to the mounts. The cancelling waveforms cannot simply be an antiphase copy of the original waveform as each shaker affects the vibration at mounts other than the one at which it is cancelling. This paper describes an approach to this multivariable control problem which measures all shaker to sensor transfer functions to give a shaker transfer function matrix, $\text{M}\text{(f)}$. The shakers are driven by voltages $\text{V}\text{(f)}$ given by $\text{V}\text{(f) = ?M}{}^{\mathrm{?1}}\text{(f)}\text{U}\text{(f)}$ where $\text{U}\text{(f)}$ is the vector of accelerations at the mounts. The controller then repeats the algorithm, this time operating on the residual accelerations at the sensors. The system is therefore adaptive and can cope with slowly changing noise spectra. Cancellations of better than ?25 dB have been achieved.     

自适应宽带有源消声

实际中存在的噪声,一般都是窄带或有色宽带噪声(简称宽带噪声),而宽带噪声更为普遍.为使宽带自适应有源消声(AANC)得到实际应用,必须保证宽带AANC系统具有良好的稳定性和较高的降噪量.为此,本文对AANC系统稳态性能作了理论分析和数值计算,得到了系统降噪量对噪声带宽、空间声传播通道、自适应滤波器参数等的依赖关系;以自由声场远场AANC为例,从声学角度对AANC系统作了物理解释,从而为改进宽带AANC系统性能提供理论依据.     

自适应空间有源消声中误差传声器的位置及个数优化

近年来对有源消声理论以及自适应有源消声技术的研究一直是相互独立着进行的,本文通过对自适应有源消声系统中误差传声器个数以及位置的优化选择,找到了它们两者之间的联系,并结合控制系统的性能特点给出了误差传声器个数及位置优化的方法,对影响自适应有源消声效果的因素做了分析讨论,实验验证了本文所得结论,并取得了良好的效果.     

An adaptive filter for gravitational-wave antennas

Summary We report on the development of an adaptive optimum filter for processing the data of a resonant bar gravitational-wave detector. This filter, based on the matched-filter theory, is adaptive in the sense that the function it realizes is derived from the actual noise spectrum of the data being analysed (instead from an idealized model of the noise). Its implementation is mostly based on frequency domain techniques. We also report on the application of the new filter to the data of the cryogenic antenna Explorer of the Rome group, with particular reference to the comparison between its performance and that of an otpimum filter with fixed values of the parameters.     

Noninvasive system identification for multichannel broadband active noise control

Many real-world applications of active noise control are characterized by transfer functions that vary significantly and unpredictably. The controller's transfer-function models must adapt to these variations. Presented here is a class of adaptive filters that accomplish quasiperiodic system identification updates for feedforward control by using blocks of input-output histories. The algorithms form a one-dimensional family linking normalized least-mean squares (LMS) adaptive filters and block recursive least-squares, termed "block projection" algorithms, and generalize the noninvasive system identification studied by Sommerfeldt and Tichy. The system identification proceeds noninvasively, producing nonparametric impulse responses. Simulations show that the algorithm's convergence is faster than that of normalized LMS, even after the additional overhead of computing the update is taken into account. Both the multichannel generalization and application of these algorithms to system identification are novel. Simulations of the algorithms' performance using measured data are presented here, while experimental results of an implemented algorithm are contained in the companion paper.     

水下充水圆柱声腔内声场自适应有源控制实验研究

本文研究了有限长充水圆柱置于水中，外声场透射形成的腔内声场自适应有源控制实验研究，结果显示，由于圆柱结构与水介质的耦合，有源控制中的声控制方法能够较好的抵消声腔主导模态和强耦合的结构主导模态，因而能够抵消在圆柱腔内较宽频带范围的声场，实验还研究了消声频带，误差传感器布放位置及肖声区域等问题。     

Adaptive wave field synthesis for broadband active sound field reproduction: signal processing

Sound field reproduction is a physical approach to the reproduction of the natural spatial character of hearing. It is also useful in experimental acoustics and psychoacoustics. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. A real reflective reproduction space thus reduces the objective accuracy of WFS. Recently, adaptive wave field synthesis (AWFS) was defined as a combination of WFS and active compensation. AWFS is based on the minimization of reproduction errors and on the penalization of departure from the WFS solution. This paper focuses on signal processing for AWFS. A classical adaptive algorithm is modified for AWFS: filtered-reference least-mean-square. This modified algorithm and the classical equivalent leaky algorithm have similar convergence properties except that the WFS solution influences the adaptation rule of the modified algorithm. The paper also introduces signal processing for independent radiation mode control of AWFS on the basis of plant decoupling. Simulation results for AWFS are introduced for free-field and reflective spaces. The two algorithms effectively reproduce the sound field and compensate for the reproduction errors at the error sensors. The independent radiation mode control allows a more flexible tuning of the algorithm.     

Optimization of the number and the positions of error sensors in adaptive active sound control

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An adaptive threshold filter for ultrasound signal rejection

The threshold filter is a frequently used technique in ultrasound B-scan to reject the small echoes contributed from backscattering that blur the tissue interface and reduce the image contrast. Note that using the threshold based on one value would simultaneously destroy local waveform features of the reflection echoes with amplitudes larger than threshold value. To resolve this problem, we developed an adaptive threshold filter based on the noise-assisted empirical mode decomposition (EMD). Computer simulations at 7.5 MHz using a single-element transducer with a bandwidth of 60% and a pulselength of 0.5 μs were carried out to explore the feasibility of the algorithm. Image measurements on the carotid artery using a 7.5 MHz, 128 elements, 1D linear array transducer with the same characteristics as those in simulations were used to verify the performance of the algorithm in practice. Compared to the result from the conventional threshold technique, the adaptive threshold filter is able to successfully suppress the smaller backscattering signals without changing the local waveform features of the preserved significant echoes due to refection.     

Multi-domain active sound control and noise shielding

This paper describes an active sound control methodology based on difference potentials. The main feature of this methodology is its ability to automatically preserve "wanted" sound within a domain while cancelling "unwanted" noise from outside the domain. This method of preservation of the wanted sounds by active shielding control is demonstrated with various broadband and realistic sound sources such as human voice and music in multiple domains in a one-dimensional enclosure. Unlike many other conventional active control methods, the proposed approach does not require the explicit characterization of the wanted sound to be preserved. The controls are designed based on the measurements of the total field on the boundaries of the shielded domain only, which is allowed to be multiply connected. The method is tested in a variety of experimental cases. The typical attenuation of the unwanted noise is found to be about 20 dB over a large area of the shielded domain and the original wanted sound field is preserved with errors of around 1 dB and below through a broad frequency range up to 1 kHz.     

Spectral shaping filter for broadband amplifiers

We present the design and implementation of a high-damage threshold spectral shaping filter that can be used to counteract gain narrowing in broadband laser amplifier systems. In contrast to earlier approaches to this problem, the filter has a wide range of tunability in both reflectivity and center wavelength. The shape of the reflectivity curve enables the production of an apodized output spectrum with a clean transform limit. We demonstrate the use of this filter in both regenerative and multipass Ti:Sapphire amplifier systems, yielding super-Gaussian amplified spectra of 70-80 nm width.     

A study of sound intensity control for active noise barriers

Active noise control systems have been applied to increase the insertion loss of noise barriers where the squared sound pressure or the total acoustic energy density is used as the cost function in previous works. The absolute value of the mean active sound intensity is chosen as the cost function to obtain extra sound insertion loss in the dark area of a hybrid active noise barrier system in this note. The strategy of minimizing the near-field sound intensity at discrete locations along the edge of the passive barrier is shown to be able to provide better far-field noise reduction than that of minimizing the squared sound pressure control. Both numerical simulations and off-line experiments are carried out with a three-channel demonstration system, where the locations of the secondary sources and the error sensors are optimized and comparisons are made between the extra sound pressure attenuation of the sound intensity control and that of the squared sound pressure control.     

双层板腔结构声传输及其有源控制研究

利用子系统模态综合方法,结合阻抗-导纳矩阵法,建立了双层板腔结构向自由空间声传输及其在入射板PZT控制、辐射板PZT控制,和腔中次级声源作动等多种控制策略下,系统物理模型的统一的分析模型,导出了系统模态响应及最优次级源强度的统一的阻抗-导纳矩阵表达式。该模型表达式各部分物理意义清晰、明确,便于进行系统耦合理论、有源控制及其机理的分析和数值研究。然后,在此基础上对双层板腔结构声传输有源控制进行了全面深入的数值计算和分析研究,重点探讨了控制方法策略及系统参数对有源控制效果的影响及其对应的控制机理。结果表明:入射板PZT作动辐射声功率最小控制策略是通过入射板、声腔和辐射板三个子系统的模态抑制或重组达到消声的目的,涉及多种复杂控制机理,对入射板、辐射板和声腔模态均有效,但对入射板模态更有效;在低频段声腔(0,0,0)模态在系统耦合响应中起主导作用,因此利用腔中次级声源作动能获得较理想的控制效果,是一种较好的控制策略;由于声腔模态与结构模态间复杂的耦合关系,使得某些频率处腔中声势能一定程度上的降低并不一定导致系统声传输损失的增加,因此,腔中声势能最小控制策略不一定能够获得理想的声传输控制效果。