MODEL-BASED SYNTHESIS AND TRANSFORMATION OF VOICED SOUNDS

Speech Communication, 1996

This paper describes the implementation of a new parametric model of the glottal geometry aimed at improving male and female speech synthesis in the framework of articulatory analysis synthesis. The model represents glottal geometry in terms of inlet and outlet area waveforms and is controlled by parameters that are tightly coupled to physiology, such as vocal fold abduction. It is embedded in an articulatory analysis synthesis system (articulatory speech mimic). To introduce naturally occurring details in our synthetic glottal flow waveforms, we modelled two different kinds of leakage: a " linked leak" and a " parallel chink". While the first is basically an incomplete glottal closure, the latter models a second glottal duct that is independent of the membranous (vibrating) part of the glottis. Characteristic for both types of leaks is that they increase dc-flow and source/tract interaction. A linked leak, however, gives rise to a steeper roll-off of the entire glottal flow spectrum, whereas a parallel chink decreases the energy of the lower frequencies more than the higher frequencies. In fact, for a parallel chink, the slope at the higher freqencies is more or less the same as in the no-leakage case. Zusammenfassung Dieser Aufsatz beschreibt die Implementierung eines neuen parametrischen Modells der glottalen Geometrie. Unsere Arbeit zielt auf eine bessere Synthese männlicher und weiblicher Sprache im Rahmen von Systemen zur artikulatorischen Analyse/Synthese. Das Modell repräsentiert die glottale Geometrie als abhängig von den Zeitfunktionen der Querschnittsflächen am Ein-und Ausgang der Glottis. Die Steuerparameter des Modells sind stark an die Physiologie angelehnt, wie zum Beispiel glottale Abduktion. Unser Modell ist Teil eines artikulatorischen Analyse/Synthese-Systems. Um die im natürlichen Vorbild vorhandenen Details in der synthetischen Zeitfunktion des glottalen Strömungs zu reproduzieren, haben wir zwei verschiedene Arten von akustisch-wirksamen glottaler Lecks (Undichtigkeiten) implementiert: ein " verbundenes Leck" und eine " parallele Spalte". Während es sich im ersten Fall im wesentlichen um einen unvollständigen glottalen Verschluss handelt, stellt der zweite Fall einen zweiten glottalen Kanal dar, der unabhängig von dem knorpeligen (vibrierenden) Teil der Glottis ist. Charaketeristisch für beide Lecktypen ist, dass sie die DC-Strömung und die Interaktion von Quellsignal und Ansatzrohr erhöhen. Ein verbundenes Leck bewirkt jedoch einen steileren Abfall des gesamten glottalen Strömungsspektrums; eine parallele Spalte hingegen erniedrigt die Energie der tieferen Frequenzen stärker als die der höheren Frequenzen. Tatsächlich ist es so, dass für eine parallele Spalte der Abfall bei höheren Frequenzen mehr oder weniger derselbe ist wie im Falle nicht vorhandener Lecks.

Electrical Engineering PhD Dissertation, Stanford …

IDENTIFICATION OF CONTROL PARAMETERS IN AN ARTICULATORY VOCAL TRACT MODEL, WITH APPLICATIONS TO THE SYNTHESIS OF SINGING Perry Raymond Cook, Stanford University, 1991 The synthesis of singing is investigated using an articulatory model ...

1994

Voice source parameters can be estimated by fitting a voice source model to the glottal flow signal which is obtained by means of inverse filtering. In this pap er we investigate the behaviour of the LF-model in a number of non-linear parameter estimation procedures. It is concluded that (1) the parameter estimates are robust against additive (white and narrow band) noise in the flow waveforms, (2) simplex search algorithms perform better than steepest descent algorithms, provided that (3) the LF-pulse is generated with an algorithm that treats all parameters as real numbers.

2007

Traditionally, voice production has been modeled using the source-filter approach. Now, the glottal airflow can be interpreted as the source and the oral and nasal cavities can be modeled as the filter. The filters are based either on representing the formants, i.e., the resonances of the vocal tract, or on modeling the articulation, i.e., the shape of the vocal tract. The source signal depends on the voice to be produced, and it is traditionally modeled as a sequence of geometric pulses or as random noise. However, more accurate source signals can be obtained by utilizing the inverse filtering technique or by modeling the vocal folds as oscillating masses.

Computer Music Journal, 1993

Medical engineering & physics, 2002

A glottal model based on physical constraints is proposed. The model describes the vocal fold as a simple oscillator, i.e. a damped mass-spring system. The oscillator is coupled with a nonlinear block, accounting for fold interaction with the airflow. The nonlinear block is modelled as a regressor-based functional with weights to be identified, and a pitch-synchronous identification procedure is outlined. The model is used to analyse voiced sounds from normal and from pathological voices, and the application of the proposed analysis procedure to voice quality assessment is discussed. 

The present article investigates into the use of the LF glottal pulse model for singing synthesis and transformation. A recent estimator of the LF-glottal pulse shape parameter (rd) is used to analyze a small collection of professional singing examples and the results are discussed in the context of recent findings relating the rd shape parameter to other speech signal parameters (intensity and vibrato). We propose a rd shape parameter model for vibrato rendering and present an algorithm that allows modifying the glottal pulse shape parameter of a given speech signal and is used to enhance the vibrato generation in a speech to singing transformation system.

The Journal of the Acoustical Society of America, 1987

Y6. Speech formant trajectory estimation using dynamic programming with modulated transition costs. David Talkin (AT&T Bell Laboratories, Room 2D-410, 600 Mountain Avenue, Murray Hill, NJ 07974) A new algorithm to track automatically speech formant frequencies have been developed. Dynamic programming is used to optimize formant trajectory estimates by imposing appropriate frequency continuity constraints. The continuity constraints are modulated by a stationarity function. The formant frequencies are selected from candidates proposed by solving for the roots of the linear predictor polynomial computed periodically from the speech waveform. The local costs of all possible mappings of the complex roots to formant frequencies are computed at each frame based on the frequencies and bandwidths of the component formants for each mapping. The cost of connecting each of these mappings with each of the mappings in the previous frame is then minimized using a modified Viterbi algorithm. Two sentences spoken by 88 males and 43 females were analyzed. The first three formants were tracked correctly in all sonorant regions in over 80% of the sentences. These performance results are based on spectrographic analysis and informal listening to formant-synthesized speech. 2:15 Y7. Some properties of autoregressive model related cepstrum. Biing-Hwang Juang and David Mansour (AT&T Bell Laboratories, 600 Mountain Avenue, Murry Hill, NJ 07974) bridging the encrypted signals in a single B-voice channel) is discussed. In particular, an enhancement procedure using a prefilter, "dither" noise and an adaptive postfilter is proposed. The system is based on an all-pole model for the degraded speech. The coefficients of the postfilter are easily derivable from the all-pole filter parameters. Informal listening tests have shown that the enhanced speech is near "toll" quality. 2:45 Y9. Invariant acoustic cues in stop consonants: A cross-language study using the Wigher distribution. H. Garudadri (Electrical Engineering,

Journal of Voice, 2007

Lehto L, Airas M, Björkner E, Sundberg J, Alku P, Comparison of two inverse filtering methods in parameterization of the glottal closing phase characteristics in different phonation types, Journal of Voice, 2007; 21 (2): 138-150.

In this paper we present a flexible deterministic plus stochastic model (DSM) approach for parametric speech analysis and synthesis with high quality. The novelty of the proposed speech processing system lies in its extended means to estimate the unvoiced stochastic component and to robustly handle the transformation of the glottal excitation source. It is therefore well suited as speech system within the context of Voice Transformation and Voice Conversion. The system is evaluated in the context of a voice quality transformation on natural human speech. The voice quality of a speech phrase is altered by means of re-synthesizing the deterministic component with different pulse shapes of the glottal excitation source. A subjective listening test suggests that the speech processing system is able to successfully synthesize and arise to a listener the perceptual sensation of different voice quality characteristics. Additionally, improvements of the speech synthesis quality compared to a baseline method are demonstrated.