This paper presents non-negative factorization of audio signals for the binaural localization of multiple sound sources within realistic and unknown sound environments. Non-negative tensor factorization (NTF) provides a sparse representation of multichannel audio signals in time, frequency, and space that can be exploited in computational audio scene analysis and robot audition for the separation and localization of sound sources. In the proposed formulation, each sound source is represented by means of spectral dictionaries, temporal activation, and its distribution within each channel (here, left and right ears). This distribution, being dependent on the frequency, can be interpreted as an explicit estimation of the Head-Related Transfer Function (HRTF) of a binaural head which can then be converted into the estimated sound source position. Moreover, the semisupervised formulation of the non-negative factorization allows us to integrate prior knowledge about some sound sources of interest whose dictionaries can be learned in advance, whereas the remaining sources are considered as background sound, which remains unknown and is estimated on the fly. The proposed NTF-based sound source localization is applied here to binaural sound source localization of multiple speakers within realistic sound environments.

In this paper, we propose a way of using the Audio-Visual Description Profile (AVDP) of the MPEG-7 standard for 2D or stereo video and multichannel audio content description. Our aim is to provide means of using AVDP in such a way, that 3D video and audio content can be correctly and consistently described. Since AVDP semantics do not include ways for dealing with 3D audiovisual content, a new semantic framework within AVDP is proposed and examples of using AVDP to describe the results of analysis algorithms on stereo video and multichannel audio content are presented.

This paper presents recent developments on our “silent speech interface ” that converts tongue and lip motions, captured by ultrasound and video imaging, into audible speech. In our previous studies, the mapping between the observed articulatory movements and the resulting speech sound was achieved using a unit selection approach. We investigate here the use of statistical mapping techniques, based on the joint modeling of visual and spectral features, using respectively Gaussian Mixture Models (GMM) and Hidden Markov Models (HMM). The prediction of the voiced/unvoiced parameter from visual articulatory data is also investigated using an artificial neural network (ANN). A continuous speech database consisting of one-hour of high-speed ultrasound and video sequences was specifically recorded to evaluate the proposed mapping techniques. Index Terms: silent speech interface, GMM, HMM, ultrasound, video, multimodal, statistical mapping

Recent improvements are presented for phonetic decoding of continuous-speech from ultrasound and optical observations of the tongue and lips in a silent speech interface application. In a new approach to this critical step, the visual streams are modeled by context-dependent multi-stream Hidden Markov Models (CD-MSHMM). Results are compared to a baseline system using context-independent modeling and a visual feature fusion strategy, with both systems evaluated on a one-hour, phonetically balanced English speech database. Tongue and lip images are coded using PCA-based feature extraction techniques. The uttered speech signal, also recorded, is used to initialize the training of the visual HMMs. Visual phonetic decoding performance is evaluated successively with and without the help of linguistic constraints introduced via a 2.5k-word decoding dictionary. Index Terms: silent speech interface, visual speech recognition, multi-stream modeling 1.

In this paper, we address the problem of audio source separation with one single sensor, using a statistical model of the sources. The approach is based on a learning step from samples of each source separately, during which we train Gaussian scaled mixture models (GSMM). During the separation step, we derive maximum a posteriori (MAP) and/or posterior mean (PM) estimates of the sources, given the observed audio mixture (Bayesian framework). From the experimental point of view, we test and evaluate the method on real audio examples.

We propose a new method to learn overcomplete dictionaries for sparse coding structured as unions of orthonormal bases. The interest of such a structure is manifold. Indeed, it seems that many signals or images can be modeled as the superimposition of several layers with sparse decompositions in as many bases. Moreover, in such dictionaries, the efficient block coordinate relaxation (BCR) algorithm can be used to compute sparse decompositions. We show that it is possible to design an iterative learning algorithm that produces a dictionary with the required structure. Each step is based on the coefficients estimation, using a variant of BCR, followed by the update of one chosen basis, using singular value decomposition. We assess experimentally how well the learning algorithm recovers dictionaries that may or may not have the required structure, and to what extent the noise level is a disturbing factor.

We propose a new method to perform the separation of two sound sources from a single sensor. This method generalizes Wiener filtering with locally stationary, non-Gaussian, parametric source models. The method involves a learning phase for which we propose three different algorithm. In the separation phase, we use a sparse non negative decomposition algorithm of our own. The algorithms are evaluated on the separation of real audio data.