Hypernasal Speech Analysis via Emperical Mode Decomposition and the Teager-Kasiser Energy Operator

In the area of speech science, one particular problem of importance has been to develop a clear method for detecting hypernasality in speech. For speech pathologists, hypernsality is a critical diagnostic used for judging the severity of velopharyngeal (nasal cavity/mouth separation) inadequacy in children with a cleft lip or cleft palate condition. For physicians and particularly neurologists, these same velopharyngeal inadequacies are believed to be linked to nervous system disorders such as Alzheimers disease and particularly Parkinson\u27s disease. One can therefore envision the need to not only find a reliable method for detecting hypernasality, but to also quantify the level (severity) of hypernasality as well. An integral component in the study of speech is the analysis of speech formants, i.e., vocal tract resonances. Traditional acoustical analysis methods of using a linear source model follow the premise that differences between normal and hypernasal speech can be distinguished by shifts or power changes in the formant frequencies and/or the widening (or narrowing) of the formant bandwidths. Such a premise, however, has not been validated with consistency. Part of the reason is that traditional acoustical analysis methods such as one-third octave band, LPC (Linear Predictive Coding), and cepstral analysis are ill-equipped to deal with the nonlinear, non-stationary, and wideband characteristics of normal and nasal speech signals. Relatively newer DSP methods that employ group delay or energy separation overcome some of these problems, but have their own issues such as possible mode mixing, noise, and the aforementioned wideband problem. However, initial investigations into energy separation methods show promise as long as these issues can be resolved. This thesis evaluates the success of a novel acoustical energy approach which deals with the mode mixing and wideband problems where: (1) a DSP sifting algorithm known as the EMD (Empirical Mode Decomposition) is first implemented to decompose the voice signal into a number of IMFs (Intrinsic Mode Functions). (2) Energy analysis is performed on each IMF via the Teager-Kaiser Energy Operator. The proposed EMD energy approach is applied to voice samples taken from the American CLP Craniofacial database and is shown to produce a clear delineation between normal and nasal samples and between different levels of hypernasality.\u2

Perceptual and acoustic impacts of aberrant properties of electrolaryngeal speech.

Thesis (Ph. D.)—Harvard-MIT Division of Health Sciences and Technology, 2003.Includes bibliographical references (p. 167-171).This electronic version was prepared by the author. The certified thesis is available in the Institute Archives and Special Collections.Ph. D

Models and analysis of vocal emissions for biomedical applications

This book of Proceedings collects the papers presented at the 4th International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications, MAVEBA 2005, held 29-31 October 2005, Firenze, Italy. The workshop is organised every two years, and aims to stimulate contacts between specialists active in research and industrial developments, in the area of voice analysis for biomedical applications. The scope of the Workshop includes all aspects of voice modelling and analysis, ranging from fundamental research to all kinds of biomedical applications and related established and advanced technologies

Articulatory-Based English Consonant Synthesis in 2-D Digital Waveguide Mesh

In articulatory speech synthesis, the 3-D shape of a vocal tract for a particular speech sound has typically been established, for example, by magnetic resonance imaging (MRI), and this is used to model the acoustic output from the tract using numerical methods that operate in either one, two or three dimensions. The dimensionality strongly affects the overall computation complexity, which has a direct bearing on the quality of the synthesized speech output. The digital waveguide mesh (DWM) is a numerical method commonly used in room acoustic modelling. A smaller space such as a vocal tract, which is about 5 cm wide and 16.5-18 cm long in adults, can also be modelled using DWM in one, two and three dimensions. The latter requires a very dense mesh requiring massive computational resources; these requirements are lessened by using a lower dimensionality (two rather than three) and/or a less dense mesh. The computational cost of 2-D digital waveguide modelling makes it a practical technique for real-time synthesis in an average PC at full (20 kHz) audio bandwidth. This research makes use of a 2-D mesh with the advantage of the availability and flexibility of existing boundary modelling and the raised-cosine impedance control to study the possibilities of using it for English consonant synthesis. The research was organized under the phonetic ‘manner’ classification of English consonants as: semi-vowel, nasal, fricative, plosive and affricate. Their production has been studied in terms of acoustic pressure wave propagation. Meshing topology was fixed to being a 4-port scattering 2-D rectilinear waveguide mesh for ease of understanding and mapping to the tract shape. As the characteristic of consonant production requires vocal tract articulation variations that are quite unlike vowels, this research adopts the articulatory trajectories using electromagnetic (mid-sagittal) articulograph (EMA) data from mngu0 to guide the change of cross-sectional vocal tract area. Generally, articulatory trajectories have been used to improve the accuracy of speech recognition and synthesis in recent decades. This research adopts the 3 trajectories to control coarticulation in consonant synthesis to demonstrate that a 2-D digital waveguide mesh (DWM) is able to simulate the formant transition accurately. The formant transitions in the results are close acoustically to natural speech and are based on controlling articulation for four places of articulation. Positions of lip, tongue tip, tongue body and tongue dorsum are inversely mapped to their corresponding cross-sectional areas. Linear interpolation between them enabled all tract movements to be modelled. The results show that tract movements are best modelled as non-linear coarticulation

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

An investigation into glottal waveform based speech coding

Coding of voiced speech by extraction of the glottal waveform has shown promise in improving the efficiency of speech coding systems. This thesis describes an investigation into the performance of such a system. The effect of reverberation on the radiation impedance at the lips is shown to be negligible under normal conditions. Also, the accuracy of the Image Method for adding artificial reverberation to anechoic speech recordings is established. A new algorithm, Pre-emphasised Maximum Likelihood Epoch Detection (PMLED), for Glottal Closure Instant detection is proposed. The algorithm is tested on natural speech and is shown to be both accurate and robust. Two techniques for giottai waveform estimation, Closed Phase Inverse Filtering (CPIF) and Iterative Adaptive Inverse Filtering (IAIF), are compared. In tandem with an LF model fitting procedure, both techniques display a high degree of accuracy However, IAIF is found to be slightly more robust. Based on these results, a Glottal Excited Linear Predictive (GELP) coding system for voiced speech is proposed and tested. Using a differential LF parameter quantisation scheme, the system achieves speech quality similar to that of U S Federal Standard 1016 CELP at a lower mean bit rate while incurring no extra delay

SINGING PORTUGUESE NASAL VOWELS: PRACTICAL STRATEGIES FOR MANAGING NASALITY IN BRAZILIAN ART SONGS

The articulation of Portuguese nasalized vowels poses some articulatory problems accompanied by negative acoustic effects for the performance of Brazilian art songs. The main objective was to find strategies that permit the singer to conciliate an idiomatic pronunciation of these vowels with a well-balanced resonance, a desirable quality in classical singing. In order to devise these strategies, the author examined sources dealing with nasalized vowels from varied perspectives: acoustic properties of vowel nasalization, phonetic and phonological aspects ofBrazilian Portuguese (BP), historical views on nasality in singing, and recent vocal pedagogy research. In addition to the overall loss of sonority, the main effect of nasalization is felt mainly in the first formant (F1) region of oral vowels, due to the introduction of nasal formants and antiformants, and to shifts in the tongue posture. Several sources report the existence of a nasality contour in BP, by which a nasalized vowel starts with an oral phase and transitions gradually to a nasal phase. The author concludes that the basic approach to sing nasalized vowels in BP is (1) to find the tongue posture corresponding to the oral vowel congener (the “core vowel”), and (2) to adjust the nasality contour in such a way that the oral portion remains prominent in order to keep the resonance balance consistent during the emission of the vowel. Once the core vowel is determined, standard vowel modification choices can be made according to voice type and the musical context in which the vowel is being sung. Some challenging excerpts from Brazilian art songs are examined, with suggestions for the application of the discussed strategies