AUDIO SCENE SEGEMENTATION USING A MICROPHONE ARRAY AND AUDITORY FEATURES

Auditory stream denotes the abstract effect a source creates in the mind of the listener. An auditory scene consists of many streams, which the listener uses to analyze and understand the environment. Computer analyses that attempt to mimic human analysis of a scene must first perform Audio Scene Segmentation (ASS). ASS find applications in surveillance, automatic speech recognition and human computer interfaces. Microphone arrays can be employed for extracting streams corresponding to spatially separated sources. However, when a source moves to a new location during a period of silence, such a system loses track of the source. This results in multiple spatially localized streams for the same source. This thesis proposes to identify local streams associated with the same source using auditory features extracted from the beamformed signal. ASS using the spatial cues is first performed. Then auditory features are extracted and segments are linked together based on similarity of the feature vector. An experiment was carried out with two simultaneous speakers. A classifier is used to classify the localized streams as belonging to one speaker or the other. The best performance was achieved when pitch appended with Gammatone Frequency Cepstral Coefficeints (GFCC) was used as the feature vector. An accuracy of 96.2% was achieved

SPEAKER AND GENDER IDENTIFICATION USING BIOACOUSTIC DATA SETS

Acoustic analysis of animal vocalizations has been widely used to identify the presence of individual species, classify vocalizations, identify individuals, and determine gender. In this work automatic identification of speaker and gender of mice from ultrasonic vocalizations and speaker identification of meerkats from their Close calls is investigated. Feature extraction was implemented using Greenwood Function Cepstral Coefficients (GFCC), designed exclusively for extracting features from animal vocalizations. Mice ultrasonic vocalizations were analyzed using Gaussian Mixture Models (GMM) which yielded an accuracy of 78.3% for speaker identification and 93.2% for gender identification. Meerkat speaker identification with Close calls was implemented using Gaussian Mixture Models (GMM) and Hidden Markov Models (HMM), with an accuracy of 90.8% and 94.4% respectively. The results obtained shows these methods indicate the presence of gender and identity information in vocalizations and support the possibility of robust gender identification and individual identification using bioacoustic data sets

Objective Estimation of Tracheoesophageal Speech Quality

Speech quality estimation for pathological voices is becoming an increasingly important research topic. The assessment of the quality and the degree of severity of a disordered speech is important to the clinical treatment and rehabilitation of patients. In particular, patients who have undergone total laryngectomy (larynx removal) produce Tracheoesophageal (TE) speech. In this thesis, we study the problem of TE speech quality estimation using advanced signal processing approaches. Since it is not possible to have a reference (clean) signal corresponding to a given TE speech (disordered) signal, we investigate in particular the non-intrusive techniques (also called single-ended or blind approaches) that do not require a reference signal to deduce the speech quality level. First, we develop a novel TE speech quality estimation based on some existing double-ended (intrusive) speech quality evaluation techniques such as the Perceptual Evaluation Speech Quality (PESQ) and Hearing Aid Speech Quality Index HASQI. The matching pursuit algorithm (MPA) was used to generate a quasi-clean speech signal from a given disordered TE speech signal. Then, by adequately choosing the parameters of the MPA (atoms, number of iterations,...etc) and using the resulting signal as our reference signal in the intrusive algorithm, we show that the resulting intrusive algorithm correlates well with the subjective scores of two TE speech databases. Second, we investigate the extraction of low complexity auditory features for the evaluation of speech quality. An 18-th order Linear Prediction (LP) analysis is performed on each voiced frame of the speech signal. Two evaluation features are extracted corresponding to higher-order statistics of the LP coefficients and the vocal tract model parameters (cross-sectional tubes areas). Using a set of 35 TE speech samples, we perform forward stepwise regression as well as K-fold cross-validation to select the best sets of features that are used in each of the regression models. Finally, the selected features are fitted to different support vector regression models yielding high correlations with subjective scores. Finally, we investigate a new approach for the estimation of the quality of TE speech using deep neural networks (DNNs). A synthetic dataset that consists of 2173 samples was used to train a DNN model that was shown to predict the TE voice quality. The synthetic dataset was formed by mixing 53 normal speech samples with modulated noise signals that had a similar envelope to the speech samples, at different speech-to-modulation noise ratios. A validated instrumental speech quality predictor was used to quantify the perceived quality of speech samples in this database, and these objective quality scores were used for training the DNN model. The DNN model was comprised of an input layer that accepted sixty relevant features extracted through filterbank and linear prediction analyses of the input speech signal, two hidden layers with 15 neurons each, and an output layer that produced the predicted speech quality score. The DNN trained on the synthetic dataset was subsequently applied to four different databases that contained speech samples collected from TE speakers. The DNN-estimated quality scores exhibited a strong correlation with the subjective ratings of the TE samples in all four databases, thus it shows strong robustness compared to those speech quality metrics developed in this thesis or those from the literature

Models and Analysis of Vocal Emissions for Biomedical Applications

The MAVEBA Workshop proceedings, held on a biannual basis, collect the scientific papers presented both as oral and poster contributions, during the conference. The main subjects are: development of theoretical and mechanical models as an aid to the study of main phonatory dysfunctions, as well as the biomedical engineering methods for the analysis of voice signals and images, as a support to clinical diagnosis and classification of vocal pathologies

Speaker identification and clustering using convolutional neural networks

Deep learning, especially in the form of convolutional neural networks (CNNs), has triggered substantial improvements in computer vision and related fields in recent years. This progress is attributed to the shift from designing features and subsequent individual sub-systems towards learning features and recognition systems end to end from nearly unprocessed data. For speaker clustering, however, it is still common to use handcrafted processing chains such as MFCC features and GMM-based models. In this paper, we use simple spectrograms as input to a CNN and study the optimal design of those networks for speaker identification and clustering. Furthermore, we elaborate on the question how to transfer a network, trained for speaker identification, to speaker clustering. We demonstrate our approach on the well known TIMIT dataset, achieving results comparable with the state of the art – without the need for handcrafted features