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The estimation of the direction-of-arrival (DOA) of one or more acoustic sources is an area that has generated much interest in recent years, with applications like automatic video camera steering and multi-party stereophonic teleconferencing entering the market. Time-difference-of-arrival (TDOA) based methods compute each relative delay using only two microphones, even though additional microphones...
In this paper we propose two novel methods for preserving the spatial information in source separation algorithms. Our approach is applicable to any source separation algorithm and is based on an additional supervised adaptive filtering with the reference signals generated by the source separation system. If a special constrained optimization scheme is applied to derive the source separation algorithm...
Spatial audio coding and enhancement address the growing commercial need to store and distribute multichannel audio and to render content optimally on arbitrary reproduction systems. In this paper, we discuss a spatial analysis-synthesis scheme which applies principal component analysis to an STFT-domain representation of the original audio to separate it into primary and ambient components, which...
This paper presents a novel solution to multichannel spatial audio coding: Spatial Squeezing Surround Audio Coding (S3AC). The S3AC scheme analyses a multichannel audio signal and downmixes it into a stereo signal pair containing both the monophonic properties of audio sources and their localization information; this avoids the need for side information. The approach uses time-frequency analysis of...
Acoustic Echo Cancellation (AEC) has become an essential and well-known enabling technology for hands-free communication and human-machine interfaces. AEC for two or more reproduction channels aims at identifying the echo paths between the microphone and each audio reproduction source in order to cancel the associated echo contribution. A number of preprocessing methods have been proposed to decorrelate...
Binaural presentation of X. Y sound is usually performed using virtual audio principles - that is, by attempting to virtually reproduce the setup of the X+Y loudspeakers in the reference room configuration. The computational cost of such playback is linear in the number of channels in the X. Y setup. We present a novel scheme that computes, offline, a spatio-temporal representation of the sound field...
Although a significant amount of research attention has been devoted to microphone-array beamforming, the performance of all the developed algorithms in practical acoustic environments is still far from meeting our expectation. So further research efforts on this topic are indispensable. In this paper, we treat a microphone array as a multiple-input multiple-output (MIMO) system and develop a general...
While current post-filtering algorithms for microphone array applications can enhance beamformer output signals, they assume that the noise is either incoherent or diffuse, and make no allowances for point noise sources which may be strongly correlated across the microphones. In this paper, we present a novel post-filtering algorithm that alleviates this assumption by tracking the spatial as well...
We present a method for simultaneous speech source separation in reverberant environments using both localization cues and a speech model. Previous source separation work has focused primarily on one or the other of these approaches; we use a novel localization cue observation noise model to allow for a natural combination of the approaches. We model speech as a Gaussian mixture model (GMM) of short-time...
Acoustic source localization in the presence of reverberation is a difficult task. Conventional approaches, based on time delay estimation performed by generalized cross correlation (GCC) on a set of microphone pairs, followed by geometric triangulation, are often unsatisfactory. Prefiltering is usually adopted to reduce the spurious peaks due to reflections. In this work an alternative strategy is...
We propose a speech separation method for a meeting situation, where each speaker sometimes speaks and the number of speakers changes every moment. Many source separation methods have already been proposed, however, they consider a case where all the speakers keep speaking: this is not always true in a real meeting. In such cases, in addition to separation, speech detection and the classification...
In this paper, first, we propose a computational-cost efficient blind source separation combining closed-form 2nd-order independent component analysis (ICA) and nonclosed-form higher-order ICA. The closed-form solution of the 2nd-order ICA has been recently presented by one of the authors. This finding motivates us to combine the closed-form 2nd-order ICA and higher-order ICA, where the preceding...
We present a study into all-pole spectral envelope estimation for the case of harmonic signals. We address the problem of the selection of the model order and propose to make use of the fact that the spectral envelope is sampled by means of the harmonic structure to derive a reasonable choice for an appropriate model order. The experimental investigation uses synthetic ARMA featured signals with varying...
This paper proposes a way of modelling the time-varying spectral energy distribution of musical instrument sounds. The model consists of an excitation signal, a body response filter, and a loss filter which implements a frequency-dependent decay. The three parts are further represented with a linear model which allows controlling the number of parameters involved. A method is proposed for estimating...
This paper describes a sound source separation method for polyphonic sound mixtures of music to build an instrument equalizer for remixing multiple tracks separated from compact-disc recordings by changing the volume level of each track. Although such mixtures usually include both harmonic and inharmonic sounds, the difficulties in dealing with both types of sounds together have not been addressed...
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