9,039 research outputs found

    Raking the Cocktail Party

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    We present the concept of an acoustic rake receiver---a microphone beamformer that uses echoes to improve the noise and interference suppression. The rake idea is well-known in wireless communications; it involves constructively combining different multipath components that arrive at the receiver antennas. Unlike spread-spectrum signals used in wireless communications, speech signals are not orthogonal to their shifts. Therefore, we focus on the spatial structure, rather than temporal. Instead of explicitly estimating the channel, we create correspondences between early echoes in time and image sources in space. These multiple sources of the desired and the interfering signal offer additional spatial diversity that we can exploit in the beamformer design. We present several "intuitive" and optimal formulations of acoustic rake receivers, and show theoretically and numerically that the rake formulation of the maximum signal-to-interference-and-noise beamformer offers significant performance boosts in terms of noise and interference suppression. Beyond signal-to-noise ratio, we observe gains in terms of the \emph{perceptual evaluation of speech quality} (PESQ) metric for the speech quality. We accompany the paper by the complete simulation and processing chain written in Python. The code and the sound samples are available online at \url{http://lcav.github.io/AcousticRakeReceiver/}.Comment: 12 pages, 11 figures, Accepted for publication in IEEE Journal on Selected Topics in Signal Processing (Special Issue on Spatial Audio

    Mirror-aided non-LOS VLC channel characterizations with a time-efficient simulation model

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    The emerging cost-efficient visible light communication (VLC)-based indoor wireless network requires an economical solution for backhaul transmission. Non-line-of-sight (non-LOS) VLC links are generally applicable candidates to set up a backhaul network without rearrangement of existing lighting fixtures. Here, we describe non-LOS channels aided by the first-order specular reflection from mirrors, which can be used to overcome the multipath effect of purely diffuse non-LOS channels. Characterizations of purely diffuse and mirror-aided non-LOS channels are conducted with a time-efficient simulation model based on an iterative algorithm. Any bounce of reflections combined with specular and diffuse reflections can be simulated using the proposed iterative VLC model in polynomial time. Simulation results show that mirror-aided non-LOS channels outperform purely diffuse non-LOS links regardless of the link configuration. The effect of concentration and directionality of non-LOS VLC channels is also shown and discussed

    Spherical harmonics based generalized image source method for simulating room acoustics

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    Allen and Berkley's image source method (ISM) is proven to be a very useful and popular technique for simulating the acoustic room transfer function (RTF) in reverberant rooms. It is based on the assumption that the source and receiver of interest are both omnidirectional. With the inherent directional nature of practical loudspeakers and the increasing use of directional microphones, the above assumption is often invalid. The main objective of this paper is to generalize the frequency domain ISM in the spherical harmonics domain such that it could simulate the RTF between practical transducers with higher-order directivity. This is achieved by decomposing transducer directivity patterns in terms of spherical harmonics and by applying the concept of image sources in spherical harmonics based propagation patterns. Therefore, from now on, any transducer can be modeled in the spherical harmonics domain with a realistic directivity pattern and incorporated with the proposed method to simulate room acoustics more accurately. We show that the proposed generalization also has an alternate use in terms of enabling RTF simulations for moving point-transducers inside pre-defined source and receiver regions.Thanks to Australian Research Council Linkage Grant funding scheme (Project No. LP160100379)

    Efficient Multiband Algorithms for Blind Source Separation

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    The problem of blind separation refers to recovering original signals, called source signals, from the mixed signals, called observation signals, in a reverberant environment. The mixture is a function of a sequence of original speech signals mixed in a reverberant room. The objective is to separate mixed signals to obtain the original signals without degradation and without prior information of the features of the sources. The strategy used to achieve this objective is to use multiple bands that work at a lower rate, have less computational cost and a quicker convergence than the conventional scheme. Our motivation is the competitive results of unequal-passbands scheme applications, in terms of the convergence speed. The objective of this research is to improve unequal-passbands schemes by improving the speed of convergence and reducing the computational cost. The first proposed work is a novel maximally decimated unequal-passbands scheme.This scheme uses multiple bands that make it work at a reduced sampling rate, and low computational cost. An adaptation approach is derived with an adaptation step that improved the convergence speed. The performance of the proposed scheme was measured in different ways. First, the mean square errors of various bands are measured and the results are compared to a maximally decimated equal-passbands scheme, which is currently the best performing method. The results show that the proposed scheme has a faster convergence rate than the maximally decimated equal-passbands scheme. Second, when the scheme is tested for white and coloured inputs using a low number of bands, it does not yield good results; but when the number of bands is increased, the speed of convergence is enhanced. Third, the scheme is tested for quick changes. It is shown that the performance of the proposed scheme is similar to that of the equal-passbands scheme. Fourth, the scheme is also tested in a stationary state. The experimental results confirm the theoretical work. For more challenging scenarios, an unequal-passbands scheme with over-sampled decimation is proposed; the greater number of bands, the more efficient the separation. The results are compared to the currently best performing method. Second, an experimental comparison is made between the proposed multiband scheme and the conventional scheme. The results show that the convergence speed and the signal-to-interference ratio of the proposed scheme are higher than that of the conventional scheme, and the computation cost is lower than that of the conventional scheme

    Audio-visual Virtual Reality System for Room Acoustics

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    We present an audio-visual Virtual Reality display system for simulated sound fields. In addition to the room acoustic simulation by means of phonon tracing and finite element method this system includes the stereoscopic visualization of simulation results using a 3D back projection system as well as auralization by use of a professional sound equipment. For auralization purposes we develop a sound field synthesis approach for accurate control of the loudspeaker system

    Blind-Matched Filtering for Speech Enhancement with Distributed Microphones

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