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A farfield random array is implemented for noise source identification. Microphone positions are optimized, with the aid of the simulated annealing method. A two-stage localization and separation algorithm is devised on the basis of the equivalent source method (ESM). In the localization stage, the active source regions are located by using the delay-and-sum method, followed by a parametric localization procedure, stochastic maximum likelihood algorithm. Multidimensional nonlinear optimization is exploited in the bearing estimation process. In the separation stage, source amplitudes are extracted by formulating an inverse problem based on the preceding source bearings identified. The number of equivalent sources is selected to be less than that of microphones to render an overdetermined problem which can be readily solved by using the Tikhonov regularization. Alternatively, the separation problem can be augmented into an underdetermined problem which can be solved by using the compressive sensing technique. Traditionally, farfield arrays only give a relative distribution of source field. However, by using the proposed method, the acoustic variables including sound pressure, particle velocity, sound intensity, and sound power can be calculated based on ESM. Numerical and experimental results of several objective and subjective tests are presented.

In this paper, four delay-and-sum (DAS) beamformers formulated in the modal domain and the space domain for open and solid spherical apertures are examined through numerical simulations. The resulting beampatterns reveal that the mainlobe of the solid spherical DAS array is only slightly narrower than that of the open array, whereas the sidelobes of the modal domain array are more significant than those of the space domain array due to the discrete approximation of continuous spherical Fourier transformation. To verify the theory experimentally, a three-dimensionally printed spherical array on which 32 micro-electro-mechanical system microphones are mounted is utilized for localization and separation of sound sources. To overcome the basis mismatch problem in signal separation, source localization is first carried out using minimum variance distortionless response beamformer. Next, Tikhonov regularization (TIKR) and compressive sensing (CS) are employed to extract the source signal amplitudes. Simulations and experiments are conducted to validate the proposed spherical array system. Objective perceptual evaluation of speech quality test and a subjective listening test are undertaken in performance evaluation. The experimental results demonstrate better separation quality achieved by the CS approach than by the TIKR approach at the cost of computational complexity.

In this paper, an evolutionary exposition is given in regard to the enhancing strategies for acoustic echo cancellers (AECs). A fixed beamformer (FBF) is utilized to focus on the near-end speaker while suppressing the echo from the far end. In reality, the array steering vector could differ considerably from the ideal freefield plane wave model. Therefore, an experimental procedure is developed to interpolate a practical array model from the measured frequency responses. Subband (SB) filtering with polyphase implementation is exploited to accelerate the cancellation process. Generalized sidelobe canceller (GSC) composed of an FBF and an adaptive blocking module is combined with AEC to maximize cancellation performance. Another enhancement is an internal iteration (IIT) procedure that enables efficient convergence in the adaptive SB filters within a sample time. Objective tests in terms of echo return loss enhancement (ERLE), perceptual evaluation of speech quality (PESQ), word recognition rate for automatic speech recognition (ASR), and subjective listening tests are conducted to validate the proposed AEC approaches. The results show that the GSC-SB-AEC-IIT approach has attained the highest ERLE without speech quality degradation, even in double-talk scenarios.

A microphone array impedance matrix measurement technique is presented for linear and passive acoustic two-port networks. Two impedance tubes fitted with three non-uniformly spaced microphones are required in the measurement. The non-uniform spacing is intended to avoid ill-posedness problems in calculating two plane-wave components traveling in opposite directions. Based on the one-port measurement, acoustic two-port networks modeled with the source and the load connected are examined. Three experimental procedures, the two-load measurement method (TLMM), the reciprocal-constrained method (RCM), and the reciprocity-symmetry-constrained method (RSCM), are developed to measure the acoustic impedance matrix. Experiments are conducted for several acoustic two-port systems to verify the proposed techniques. The results demonstrate the efficacy of the three experimental procedures when applied to symmetrical and reciprocal systems. For asymmetrical systems, the TLMM and RCM are preferred over the RSCM for measuring the impedance matrix. On top of that, the non-uniform array in conjunction with TLMM is extended to a general electroacoustic two-port system, which can be regarded as a unique contribution of the present work.

This paper presents a new design and implementation of a balanced armature speaker (BAS), which is composed of permanent magnetic circuits, a moving armature, and a coil. The armature rocks about a pivot with the coil at one end and the permanent magnet on another. A magnetic circuit analysis is conducted for the designed BAS to formulate the force factor, which is required for modeling the coupling between the electrical and mechanical systems. In addition, an electromechanoacoustical analogous circuit is established for the BAS, which bears the same structure as the moving coil loudspeaker, except that the force factor is different. A hybrid model, which combines the lumped parameter model in the electrical and acoustical domains with a finite element model in the mechanical domain, is developed to model the high-frequency response because of the high-order modes of the membrane, the drive rod, and the armature. The electroacoustic analysis is experimentally verified. The results indicate that the sound pressure response that is simulated using the hybrid model is in superior agreement with the measured response to that simulated using the lumped parameter model.

A shotgun microphone is a highly directional pickup device widely used in noisy environments. The key element that leads to its superior directivity is a tube with multiple slot openings along its length. One traditional way to model the directional response of a shotgun is to assume plane waves traveling in the tube as if it is in the free field. However, the frequency response and directivity predicted by this traveling wave model can differ drastically from practical measurements. In this paper, an in-depth electroacoustic analysis was conducted to examine the problem by considering the standing waves inside the tube with an analogous circuit containing phased pressure sources and T-networks of tube segments. A further refinement is to model the housing diffraction effect with the aid of the equivalent source method (ESM). The on-axis frequency response and directivity pattern predicted by the proposed model are in close agreement with the measurements. From the results, a peculiar bifurcation phenomenon of directivity pattern at the Helmholtz frequency was also noted. While the shotgun behaves like an endfire array above the Helmholtz frequency, it becomes a broadside array below the Helmholtz frequency. The standing wave effect can be mitigated by covering the slot openings with mesh screen, which was found to alter the shotgun response to be closer to that of the traveling wave model above a critical frequency predicted by the half-wavelength rule. A mode-switching model was developed to predict the directional responses of mesh-treated shotguns.

This study demonstrates a SOI condenser MEMS microphone consisting of planar interdigitated sensing electrodes, deformable diaphragm, and back chamber. No back-plate is required for this design, thus, the acoustic impedance is reduced. The merits of this study are: (1) the back-plate is replaced by planar interdigitated sensing electrodes to prevent in-use pull-in and process stiction between diaphragm and back-plate, (2) out-of-plane area-changing capacitive sensing provides a better linearity to sound pressure variation, (3) large and flat diaphragm implemented by SOI and DRIE process increases the acoustic sensitivity. The proposed microphone with 600µm diameter diaphragm and 42 pairs sensing electrodes is realized and tested. Measurements show a flat response between 1 kHz to 20 kHz. The sensitivity of microphone is −60.1dB (ref. 1V/1Pa) at 1 kHz.