Signal subspace has been widely exploited to localize multiple speech sources. However, most signal subspace methods cannot count the number of sources, and do not make use of speech sparsity in the frequency domain. This paper presents a grid search window-dominant signal subspace (GS-WDSS) method and a closed-form WDSS (CF-WDSS) method to localize short-term speech sources. Such methods are based upon the generalized sparsity assumption that each window containing some time-adjacent bins is dominated by one source, as opposed to the conventional assumption that each individual bin is dominated by one source. The generalized assumption enables the principal eigenvector of the spatial correlation matrix on each window to span the signal subspace of the window-dominant source. The direction-of-arrival (DOA) of the dominant source is estimated from the principal eigenvector. The DOAs and the number of sources are eventually summarized from the DOA histogram of all dominant sources. The conventional assumption is a special case of the generalized assumption. By using the generalized assumption, the performance in estimating DOAs of the window-dominant sources is significantly improved at the cost of acceptable masking effect. The superiority of the proposed methods is verified by simulated and real experiments.

Spatial aliasing and spatial resolution are the two issues faced by most multiple speech source localization methods. The histogram of time delays is a simple but effective method to deal with these two issues on linear arrays. But few methods were capable of applying the time delay histogram to directional-of-arrivals (DOAs) estimation using a planar array. This paper proposes a novel method to estimate DOAs of multiple speech sources based on time delay histograms across all microphones of a planar array. The pairwise time delays of different sources are firstly obtained from each time delay histogram, and then, the time delays are identified with variant speech sources. Eventually, the DOA of each source is estimated by regression over its associated time delays. We conducted some experiments in both simulated and real environments to evaluate the proposed method using an eight-element circular array. The experimental results confirmed not only its high computational efficiency, but also its superiority in spatial resolution and spatial anti-aliasing.

Phase difference regression (PDR) was widely utilized to estimate Direction-of-Arrival (DOA) for linear arrays because of its high time resolution and high computational efficiency. However, conventional regression methods were seldom reported to estimate DOA using planar arrays. This paper proposes a regression method to derive DOA from all phase differences on all frequencies for a planar array. The DOA is represented as the function of the array topology and phase differences between all microphones. Moreover, the proposed method considers another two problems that were often ignored by most regression methods. One is the problem about the period of phase difference in the regression cost function. The other is the signal enhancement that can effectively suppress the acoustic interference. We conducted some experiments in simulated environment to evaluate the proposed method using a 9-element circular array. The experimental results confirmed its superiority in both computational efficiency and robustness. Index Terms: Planar array, phase difference regression, signal enhancement, sound source localization.

Heavy computational load and acoustic interferences are two major problems to speech source localization in real applications. Conventional methods can mitigate one problem, but deteriorate the other. This letter proposes an algorithm of direction-of-arrival (DOA) estimation, which is both computationally efficient and robust in the presence of acoustic interferences. The robustness is considered in two aspects. One is the eigenanalysis-based enhancement to reduce acoustic interferences such as noise and reverberation. The other is the coefficients that weight the pairwise time delays to mitigate the effect of delay outliers on DOA. The high computational efficiency is achieved by making use of a concave cost function, from which, the optimal estimate of DOA is given by a closed-form solution. The grid-search method often adopted in conventional algorithms is no longer used in this algorithm. We conduct some experiments in both simulated and real environments with a 9-element circular array. The proposed algorithm runs about ten times faster than Steered Response Power PHAse Transform (SRP-PHAT), and outperforms SRP-PHAT in terms of robustness.

The temporal correlation of speech presence/absence is widely used in noise estimation. The most popular technique for exploiting temporal correlation is the smoothing of noisy spectra using a time-recursive filter, in which the forgetting factor is controlled by speech presence probability. However, this technique is not unified into a theoretical framework that enables optimal noise estimation. In theory, hidden Markov models (HMMs) are superior to this technique in modeling temporal correlation. HMMs can model a time sequence of presence/absence of speech signal as a dynamic process of the transition between speech and non-speech states. Moreover, a number of methods, such as maximum likelihood, are available for optimal estimation of HMM parameters. This paper presents a constrained sequential HMM for modeling the log-power sequence on each frequency band. The emission probability of each HMM state is represented by a Gaussian model. The Gaussian mean of the non-speech state is considered as the optimal estimate of noise logarithmic power. The HMM parameter set is sequentially estimated from one frame to another on the basis of maximum likelihood. The proposed method is compared with well-established algorithms through various experiments. Our method delivers more accurate results and does not rely on the assumption of the “non-speech signal onset” as do most algorithms.

How to utilize the time correlation of speech/nonspeech presence is a crucial problem faced by noise estimators. The popular technique of exploiting such correlation is to smooth noisy spectra by using a temporal recursive filter with a time-varying smoothing factor. But this technique cannot warrant the statistical optimality. In theory, hidden Markov model (HMM) is more desirable than this technique. It can give an elaborate description of speech/nonspeech transition. Moreover, some theoretical frameworks, such as maximum likelihood (ML), are available for optimal estimation. This paper presents a constrained sequential HMM to model the time correlation of speech/nonspeech presence of an individual log-power sequence. Its parameter set is on-line adapted to varying signals based on a ML framework. We compared its performance with that of well-established algorithms by speech enhancement experiments. The results confirmed its promising performance.

How to construct models for speech/nonspeech discrimination is a crucial point for voice activity detectors (VADs). Semi-supervised learning is the most popular way for model construction in conventional VADs. In this correspondence, we propose an unsupervised learning framework to construct statistical models for VAD. This framework is realized by a sequential Gaussian mixture model. It comprises an initialization process and an updating process. At each subband, the GMM is firstly initialized using EM algorithm, and then sequentially updated frame by frame. From the GMM, a self-regulatory threshold for discrimination is derived at each subband. Some constraints are introduced to this GMM for the sake of reliability. For the reason of unsupervised learning, the proposed VAD does not rely on an assumption that the first several frames of an utterance are nonspeech, which is widely used in most VADs. Moreover, the speech presence probability in the time-frequency domain is a byproduct of this VAD. We tested it on speech from TIMIT database and noise from NOISEX-92 database. The evaluations effectively showed its promising performance in comparison with VADs such as ITU G.729B, GSM AMR, and a typical semi-supervised VAD.