Huseyin Hacihabiboglu

On the Design and Implementation of Higher Order Differential Microphones

A novel systematic approach to the design of directivity patterns of higher order differential microphones is proposed. The directivity patterns are obtained by optimizing a cost function which is a convex combination of a front-back energy ratio and uniformity within a frontal sector of interest. Most of the standard directivity patterns - omnidirectional, cardioid, subcardioid, hypercardioid, supercardioid - are particular solutions of this optimization problem with specific values of two free parameters: the angular width of the frontal sector and the convex combination factor. More general solutions of practical use are obtained by varying these two parameters. Many of these optimal directivity patterns are trigonometric polynomials with complex roots. A new differential array structure that enables the implementation of general higher order directivity patterns, with complex or real roots, is then proposed. The effectiveness of the proposed design framework and the implementation structure are illustrated by design examples, simulations, and measurements.

Intensity vector direction exploitation for exhaustive blind source separation of convolutive mixtures

This article presents a technique that uses the intensity vector direction exploitation (IVDE) method for exhaustive separation of convolutive mixtures. While only a four-element compact sensor array is used, multiple channels for all possible source directions are produced by exhaustive separation. Singular value decomposition (SVD) is then applied to determine the signal subspace and the directions of the local maxima of the signal energy. This information is then used to select the channels containing the individual sources. While the original IVDE method requires the prior knowledge of the directions of sources for separation, the present method eliminates this need and achieves fully blind separation. Performing SVD at a post-processing stage also improves the sound quality. The method has been tested for convolutive mixtures of up to four sources and typical separation performances are given.

Automatic Speech Emotion Recognition Using Auditory Models with Binary Decision Tree and SVM

Affective computing is a term for the design and development of algorithms that enable computers to recognize the emotions of their users and respond in a natural way. Speech, along with facial gestures, is one of the primary modalities with which humans express their emotions. While emotional cues in speech are available to an interlocutor in a dyadic conversation setting, their subjective recognition is far from accurate. This is due to the human auditory system which is primarily non-linear and adaptive. An automatic speech emotion recognition algorithm based on a computational model of the human auditory system is described in this paper. The devised system is tested on three emotional speech datasets. The results of a subjective recognition task is also reported. It is shown that the proposed algorithm provides recognition rates that are comparable to those of human raters.

Panoramic recording and reproduction of multichannel audio using a circular microphone array

Multichannel audio reproduction generally suffers from one or both of the following problems: i) the recorded audio has to be artificially manipulated to provide the necessary spatial cues, which reduces the consistency of the reproduced sound field with the actual one, and ii) reproduction is not panoramic, which degrades realism when the listener is not seated in a desired ideal position facing the center channel. A recording method using a circularly symmetric array of differential microphones, and a reproduction method using a corresponding array of loudspeakers is presented in this paper. Design of microphone directivity patterns to achieve a panoramic auditory scene is discussed. Objective results in the form of active intensity diagrams are presented.

Analysis and Design of Multichannel Systems for Perceptual Sound Field Reconstruction

This paper presents a systematic framework for the analysis and design of circular multichannel surround sound systems. Objective analysis based on the concept of active intensity fields shows that for stable rendition of monochromatic plane waves it is beneficial to render each such wave by no more than two channels. Based on that finding, we propose a methodology for the design of circular microphone arrays, in the same configuration as the corresponding loudspeaker system, which aims to capture inter-channel time and intensity differences that ensure accurate rendition of the auditory perspective. The methodology is applicable to regular and irregular microphone/speaker layouts, and a wide range of microphone array radii, including the special case of coincident arrays which corresponds to intensity-based systems. Several design examples, involving first and higher-order microphones are presented. Results of formal listening tests suggest that the proposed design methodology achieves a performance comparable to prior art in the center of the loudspeaker array and a more graceful degradation away from the center.

Time-Domain Simulation of Directive Sources in 3-D Digital Waveguide Mesh-Based Acoustical Models

Digital waveguide mesh (DWM) models offer a simple, accurate, time-domain, numerical solution of the wave equation. A specific case where such accurate and computationally simple solutions are needed is the acoustical modeling of open or closed volumes. It is possible to model 3-D propagation of waves in enclosures such as rooms using DWM models. Generally, idealized omnidirectional sources are used for obtaining the impulse response in the DWM. However, real-life sound sources are never completely isotropic, causing wavefronts with directional properties. This paper presents two methods to simulate analytical and empirical directivities in 3-D DWM models in the far-field. The first method is based on the direct excitation of the mesh with the velocity component of the directional source and is used to simulate analytical sources. The second method is based on the weighting of velocity components generated by an omnidirectional source at different octave-bands and is used to simulate sources with frequency-dependent empirical directivity functions. A simple interpolation method for obtaining a closed-form description of the directivity function from incomplete directivity data is also proposed. Simulation results are presented for two sources in an acoustical model of a rectangular room.

Head-related transfer function filter interpolation by root displacement

Head-related transfer function (HRTF) filters are used in virtual auditory displays for the binaural synthesis of the direction of a sound source over headphones. Once low-order HRTF filters are designed, the interpolation of these filters becomes an important issue for the synthesis of moving sound sources. An HRTF filter interpolation method based on the displacement of HRTF filter roots is proposed. It is possible to obtain a minimum-phase interpolated filter given that the original filters are also minimum-phase. The computational complexity of the method is the lower than that of the linear interpolation of magnitude responses.

Theoretical Analysis of Open Spherical Microphone Arrays for Acoustic Intensity Measurements

Acoustic intensity is a vectorial measure of acoustic energy flow through a given region of interest. Three-dimensional measurement of acoustic intensity requires special microphone array configurations. This paper provides a theoretical analysis of open spherical microphone arrays for the 3-D measurement of acoustic intensity. The calculations of the pressure and the particle velocity components of the sound field inside a closed volume are expressed using the Kirchhoff-Helmholtz integral equation. The conditions which simplify the calculation are identified. This calculation is then constrained to a finite set of microphones positioned at prescribed points on an open sphere. Several open spherical array topologies are proposed. Their magnitude and directional errors and measurement bandwidths are investigated via numerical simulations. A comparison with conventional open-sphere 3-D intensity probes is presented.

Morphing of transient sounds based on shift-invariant discrete wavelet transform and singular value decomposition

In this paper, a new morphing algorithm for transient sounds is introduced. Input sounds are first projected onto orthogonal bases from which intermediate or novel sounds can be generated. The proposed algorithm uses a shift invariant version of discrete wavelet transform and the singular value decomposition (SVD) to represent the input sound signals over a set of orthogonal bases. Interpolation is carried out between the weight vectors from the SVD to produce a new weight vector used for synthesising a new set of wavelet coefficients. The morphed sound is generated by taking the inverse discrete wavelet transform of the combined weighted bases. The proposed algorithm not only generates a range of new sounds, but also represents the input sounds in a more compact fashion.

Blind source separation and directional audio synthesis for binaural auralization of multiple sound sources using microphone array recordings

Microphone array signal processing techniques are extensively used for sound source localisation, acoustical characterisation and sound source separation, which are related to audio analysis. However, the use of microphone arrays for auralisation, which is generally related to synthesis, has been limited so far. This paper proposes a method for binaural auralisation of multiple sound sources based on blind source separation (BSS) and binaural audio synthesis. A BSS algorithm is introduced that exploits the intensity vector directions in order to generate directional signals. The directional signals are then used in the synthesis of binaural recordings using head related transfer functions. The synthesised recordings subsume the indirect information about the auditory environment conveying the source positions and the acoustics similar to dummy head recordings. Test recordings were made with a compact microphone array in two different indoor environments. Original and synthesized binaural recordings were compared by informal listening tests.