In the development process of noise-reduction algorithms, an objective machine-driven intelligibility measure which shows high correlation with speech intelligibility is of great interest. Besides reducing time and costs compared to real listening experiments, an objective intelligibility measure could also help provide answers on how to improve the intelligibility of noisy unprocessed speech. In this paper, a short-time objective intelligibility measure (STOI) is presented, which shows high correlation with the intelligibility of noisy and time-frequency weighted noisy speech (e.g., resulting from noise reduction) of three different listening experiments. In general, STOI showed better correlation with speech intelligibility compared to five other reference objective intelligibility models. In contrast to other conventional intelligibility models which tend to rely on global statistics across entire sentences, STOI is based on shorter time segments (386 ms). Experiments indeed show that it is beneficial to take segment lengths of this order into account. In addition, a free Matlab implementation is provided.

An Algorithm for Intelligibility Prediction of Time–Frequency Weighted Noisy Speech

The theory of recording and reproduction of three-dimensional sound fields based on spherical harmonics is reviewed and extended. Free-field, sphere, and general recording arrays are reviewed, and the mode-matching and simple source approaches to sound reproduction in anechoic environments are discussed. Both methods avoid the need for both monopole and dipole loudspeakers—as required by the Kirchhoff–Helmholtz integral. An error analysis is presented and simulation examples are given. It is also shown that the theory can be extended to sound reproduction in reverberant environments.

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Three-Dimensional Surround Sound Systems Based on Spherical Harmonics

Transfer-function measurements using sweeps as excitation signals rather than pseudonoise signals show significantly higher immunity against distortion and time variance. Capturing binaural room impulse responses for high-quality auralization purposes requires a signal-to-noise ratio (SNR) greater than 90 dB, which is unattainable with maximum-length sequence (MLS) measurements because of loudspeaker nonlinearity, but it is fairly easy to reach with sweeps due to the possibility of complete rejection of harmonic distortion. Before investigating the differences and practical problems of measurements with MLS and sweeps and arguing why sweeps are the preferable choice for the majority of measurement tasks, the existing methods of obtaining transfer functions are reviewed. The continual need to use preemphasized excitation signals in acoustical measurements is also addressed. A method to create sweeps with arbitrary spectral contents, but constant or prescribed frequency-dependent temporal envelope, is presented. Finally, the possibility of simultaneously analyzing transfer functions and harmonics is investigated.

Transfer-Function Measurement with Sweeps

The spatialization of the sound field in a room is studied, in particular the evolution of room impulse responses as a function of their spatial positions. It was observed that the multidimensional spectrum of the solution of the wave equation has an almost bandlimited character. Therefore, sampling and interpolation can easily be applied using signals on an array. The decay of the spectrum is studied on both temporal and spatial frequency axes. The influence of the decay on the performance of the interpolation is analyzed. Based on the support of the spectrum, the number and the spacing between the microphones is determined for the reconstruction of the sound pressure field up to a certain temporal frequency and with a certain reconstruction quality. The optimal sampling pattern for the microphone positions is given for the linear, planar and three-dimensional case. Existing techniques usually make use of room models to recreate the sound field present at some point in the space. The presented technique simply starts from the measurements of the sound pressure field in a finite number of positions and with this information the sound pressure field can be recreated at any spatial position. Finally, simulations and experimental results are presented and compared with the theory

/pdf/the-plenacoustic-function-and-its-sampling-4sab1ik208.pdf

The Plenacoustic Function and Its Sampling

A sound system for capturing and reproducing sounds produced by a plurality of sound sources. The system comprises a device for receiving sounds produced by the plurality of sound sources and converting the separately received sounds to a plurality of separate audio signals without mixing the audio signals. The system may further comprise a device for separately storing the plurality of separate audio signals on a recording medium without mixing the audio signals and a device for reading the stored audio signals from the recording medium. A sound system and method for modeling a sound field generated by a sound source and creating a sound event based on the modeled sound field is also disclosed. The system and method captures a sound field over an enclosing surface, models the sound field and enables reproduction of the modeled sound field.

System and method for integral transference of acoustical events

The concept of wave field synthesis (WFS) was introduced by Betkhout in 1988. It enables the generation of sound fields with natural temporal and spatial properties within a volume or area bounded by arrays of loudspeakers. Applications are found in real time performances as well as in reproduction of multitrack recordings. A logic next step was the formulation of a new wave field analysis (WFA) concept by Berkhout in 1997, where sound fields in enclosures are recorded with arrays of microphones and analyzed with postprocessing techniques commonly used in acoustical imaging. This way, both the temporal and spatial properties of the sound field can be investigated and understood. WFS and WFA meet in auralization applications: sound fields measured (or modeled) along arrays of microphone positions can be generated by arrays of loudspeakers for perceptual evaluation.

/pdf/wave-field-synthesis-and-analysis-using-array-technology-22row5fx2m.pdf

Wave field synthesis and analysis using array technology

A new method is proposed to acquire impulse responses in concert halls with large signal‐to‐noise ratios and with a high resolution. In our proposal an omnidirectional loudspeaker is used which is driven by an amplified sweep: a signal containing all frequencies of interest smeared out in time. By using a deconvolution technique, an almost perfect pulse is obtained with a high peak pressure and a short effective duration. Measurements were made in two different concert halls to illustrate the practical implications of the new technique.

A new method to acquire impulse responses in concert halls

A new concept for multichannel loudspeakers is introduced for application in wave field synthesis (WFS). It is a multi-actuator panel (MAP), which consists of a damped acoustic radiation panel with a number of exciters that are used to generate the WFS wave field. It is first shown that distributed-mode loudspeaker (DML) technology can be applied successfully to WFS. There are, however, some drawbacks to apply standard DML panels for this application. This led to the development of the MAPs. It is shown from theory and is confirmed by measurements that these panels can be designed in such a way that they are ideally suited for WFS sound reproduction.

Multi-Actuator Panels (MAPs) as Loudspeaker Arrays for Wave Field Synthesis

Wave Field Synthesis is a method of sound reproduction, based on the precise construction of the desired wave field by secondary sources, implemented as arrays of loudspeakers. The method is derived from acoustic wave field theory, and implemented in a practical approach based on physical as well as perceptual laws. An overview is given of the theory, the implementation as a Laboratory Demonstration System and some interesting applications of the method. Presently, the method is further developed in an ISTproject of the European Commission, called CARROUSO.

Acoustic rendering with wave field synthesis

Spatial reproduction of the voices of conference participants can greatly enhance the performance of a life-size videoconferencing system in terms of qualities such as speech intelligibility, speaker identification and more generally the naturalness of a conference. A very suitable technique to implement accurate spatial sound reproduction including depth is Wave Field Synthesis (WFS). This paper presents results of research that has been carried out to investigate the combination of WFS with 2D video projection, including subjective experiments on sound localization, correspondence of perceived auditory and visual source directions and speaker identification in situations with multiple speakers, as well as speech intelligibility tests, investigations on the applicability of Distributed Mode Loudspeakers in WFS and coloration artifacts due to discretization of the loudspeaker array.

Marinus M. Boone

Papers

Wave field synthesis and analysis using array technology

A new method to acquire impulse responses in concert halls

Multi-Actuator Panels (MAPs) as Loudspeaker Arrays for Wave Field Synthesis

Acoustic rendering with wave field synthesis

Application of Wave Field Synthesis in Life-size Videoconferencing