Inverse filtering of room acoustics
01 Feb 1988-IEEE Transactions on Acoustics, Speech, and Signal Processing (IEEE)-Vol. 36, Iss: 2, pp 145-152
TL;DR: In this article, a novel method is proposed for realizing exact inverse filtering of acoustic impulse responses in room, based on the principle called the multiple-input/output inverse theorem (MINT).
Abstract: A novel method is proposed for realizing exact inverse filtering of acoustic impulse responses in room. This method is based on the principle called the multiple-input/output inverse theorem (MINT). The inverse is constructed from multiple finite-impulse response (FIR) filters (transversal filters) by adding some extra acoustic signal-transmission channels produced by multiple loudspeakers or microphones. The coefficients of these FIR filters can be computed by the well-known rules of matrix algebra. Inverse filtering in a sound field is investigated experimentally. It is shown that the proposed method is greatly superior to previous methods that use only one acoustic signal-transmission channel. The results prove the possibility of sound reproduction and sound reception without any distortion caused by reflected sounds. >
Citations
More filters
[...]
TL;DR: The importance of having a clear understanding of the principles behind both the acoustics and the electrical control in order to appreciate the advantages and limitations of active noise control is emphasized.
Abstract: Active noise control exploits the long wavelengths associated with low frequency sound. It works on the principle of destructive interference between the sound fields generated by the original primary sound source and that due to other secondary sources, acoustic outputs of which can be controlled. The acoustic objectives of different active noise control systems and the electrical control methodologies that are used to achieve these objectives are examined. The importance of having a clear understanding of the principles behind both the acoustics and the electrical control in order to appreciate the advantages and limitations of active noise control is emphasized. A brief discussion of the physical basis of active sound control that concentrates on three-dimensional sound fields is presented. >
965 citations
Journal Article•
TL;DR: The theory of recording and reproduction of three-dimensional sound fields based on spherical harmonics is reviewed and extended in this paper, where mode-matching and simple source approaches to sound reproduction in anechoic environments are discussed.
Abstract: 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.
467 citations
Book•
28 Jul 2010
TL;DR: Speech Dereverberation presents the most important current approaches to the problem of reverberation and defines the current state of the art and encourages further work on this topic by offering open research questions to exercise the curiosity of the reader.
Abstract: Speech dereverberation is a signal processing technique of key importance for successful hands-free speech acquisition in applications of telecommunications and automatic speech recognition. Over the last few years, speech dereverberation has become a hot research topic driven by consumer demand, the availability of terminals based on Skype which encourage hands-free operation and the development of promising signal processing algorithms. Speech Dereverberation gathers together an overview, a mathematical formulation of the problem and the state-of-the-art solutions for dereverberation. Speech Dereverberation presents the most important current approaches to the problem of reverberation. It begins by providing a focused and digestible review of the relevant topics in room acoustics and also describes key performance measures for dereverberation. The algorithms are then explained together with relevant mathematical analysis and supporting examples that enable the reader to see the relative strengths and weaknesses of the various techniques, as well as giving a clear understanding of the open questions still to be addressed in this topic. Techniques rooted in speech enhancement are included, in addition to a substantial treatment of multichannel blind acoustic system identification and inversion. The TRINICON framework is shown in the context of dereverberation to be a powerful generalization of the signal processing for a important range of analysis and enhancement techniques. Speech Dereverberation offers the reader an overview of the subject area, as well as an in-depth text on the advanced signal processing involved. The book benefits the reader by providing such a wealth of information in one place, defines the current state of the art and, lastly, encourages further work on this topic by offering open research questions to exercise the curiosity of the reader. It is suitable for students at masters and doctoral level, as well as established researchers.
398 citations
TL;DR: NDLP can robustly estimate an inverse system for late reverberation in the presence of noise without greatly distorting a direct speech signal and can be implemented in a computationally efficient manner in the time-frequency domain.
Abstract: This paper proposes a statistical model-based speech dereverberation approach that can cancel the late reverberation of a reverberant speech signal captured by distant microphones without prior knowledge of the room impulse responses. With this approach, the generative model of the captured signal is composed of a source process, which is assumed to be a Gaussian process with a time-varying variance, and an observation process modeled by a delayed linear prediction (DLP). The optimization objective for the dereverberation problem is derived to be the sum of the squared prediction errors normalized by the source variances; hence, this approach is referred to as variance-normalized delayed linear prediction (NDLP). Inheriting the characteristic of DLP, NDLP can robustly estimate an inverse system for late reverberation in the presence of noise without greatly distorting a direct speech signal. In addition, owing to the use of variance normalization, NDLP allows us to improve the dereverberation result especially with relatively short (of the order of a few seconds) observations. Furthermore, NDLP can be implemented in a computationally efficient manner in the time-frequency domain. Experimental results demonstrate the effectiveness and efficiency of the proposed approach in comparison with two existing approaches.
371 citations
TL;DR: This work combines cross-power minimization of second-order source separation with geometric linear constraints used in adaptive beamforming to resolve some of the ambiguities inherent in the independence criterion such as frequency permutations and degrees of freedom provided by additional sensors.
Abstract: Convolutive blind source separation and adaptive beamforming have a similar goal-extracting a source of interest (or multiple sources) while reducing undesired interferences. A benefit of source separation is that it overcomes the conventional cross-talk or leakage problem of adaptive beamforming. Beamforming on the other hand exploits geometric information which is often readily available but not utilized in blind algorithms. We propose to join these benefits by combining cross-power minimization of second-order source separation with geometric linear constraints used in adaptive beamforming. We find that the geometric constraints resolve some of the ambiguities inherent in the independence criterion such as frequency permutations and degrees of freedom provided by additional sensors. We demonstrate the new method in performance comparisons for actual room recordings of two and three simultaneous acoustic sources.
341 citations
References
More filters
TL;DR: The theoretical and practical use of image techniques for simulating the impulse response between two points in a small rectangular room, when convolved with any desired input signal, simulates room reverberation of the input signal.
Abstract: Image methods are commonly used for the analysis of the acoustic properties of enclosures. In this paper we discuss the theoretical and practical use of image techniques for simulating, on a digital computer, the impulse response between two points in a small rectangular room. The resulting impulse response, when convolved with any desired input signal, such as speech, simulates room reverberation of the input signal. This technique is useful in signal processing or psychoacoustic studies. The entire process is carried out on a digital computer so that a wide range of room parameters can be studied with accurate control over the experimental conditions. A fortran implementation of this model has been included.
3,720 citations
TL;DR: In this article, the authors used a point image method to solve for wall reflections and a Nyquist plot was used to determine whether a given room impulse response was minimum phase when the initial delay was removed.
Abstract: When a conversation takes place inside a room, the acoustic speech signal is distorted by wall reflections. The room’s effect on this signal can be characterized by a room impulse response. If the impulse response happens to be minimum phase, it can easily be inverted. Synthetic room impulse responses were generated using a point image method to solve for wall reflections. A Nyquist plot was used to determine whether a given impulse response was minimum phase. Certain synthetic room impulse responses were found to be minimum phase when the initial delay was removed. A minimum phase inverse filter was successfully used to remove the effect of a room impulse response on a speech signal.
377 citations
01 Jan 1984
TL;DR: The proposed adaptive inverse modeling process is a promising new approach to the design of adaptive control systems and can be used to obtain a stable controller, whether the plant is minimum or non-minimum phase.
Abstract: A few of the well established methods of adaptive signal processing are modified and extended for application to adaptive control. An unknown plant will track an input command signal if the plant is preceded by a controller whose transfer function approximates the inverse of the plant transfer function. An adaptive inverse modeling process can be used to obtain a stable controller, whether the plant is minimum or non-minimum phase. No direct feedback is involved. However the system output is monitored and utilized in order to adjust the parameters of the controller. The proposed method is a promising new approach to the design of adaptive control systems.
57 citations