Cantor Digitalis is a performative singing synthesizer that is composed of two main parts: a chironomic control interface and a parametric voice synthesizer. The control interface is based on a pen/touch graphic tablet equipped with a template representing vocalic and melodic spaces. Hand and pen positions, pen pressure, and a graphical user interface are assigned to specific vocal controls. This interface allows for real-time accurate control over high-level singing synthesis parameters. The sound generation system is based on a parametric synthesizer that features a spectral voice source model, a vocal tract model consisting of parallel filters for vocalic formants and cascaded with anti-resonance for the spectral effect of hypo-pharynx cavities, and rules for parameter settings and source/filter dependencies between fundamental frequency, vocal effort, and formants. Because Cantor Digitalis is a parametric system, every aspect of voice quality can be controlled (e.g., vocal tract size, aperiodicities in the voice source, vowels, and so forth). It offers several presets for different voice types. Cantor Digitalis has been played on stage in several public concerts, and it has also been proven to be useful as a tool for voice pedagogy. The aim of this article is to provide a comprehensive technical overview of Cantor Digitalis.
Whisper-to-speech conversion is motivated by laryngeal disorders, in which malfunction of the vocal folds leads to loss of voicing. Many patients with laryngeal disorders can still produce functional whispers, since these are characterised by the absence of vocal fold vibration. Whispers therefore constitute a common ground for speech rehabilitation across many kinds of laryngeal disorder. Whisper-to-speech conversion involves recreating natural-sounding speech from recorded whispers, and is a non-invasive and non-surgical rehabilitation that can maintain a natural method of speaking, unlike the existing methods of rehabilitation. This paper proposes a new rule-based method for whisper-to-speech conversion that replaces the noisy whisper sound source with a synthesised speech-like harmonic source, while maintaining the vocal tract component unaltered. In particular, a novel glottal source generator is developed in which whisper information is used to parameterise the excitation through a high-quality glottis model. Evaluation of the system against the standard pulse train excitation method reveals significantly improved performance. Since our method is glottis-based, it is potentially compatible with the many existing vocal tract component adaptation systems.
The estimation of glottal flow from a speech waveform is a key method for speech analysis and parameterization. Significant research effort has been made to dissociate the first vocal tract resonance from the glottal formant (the low-frequency resonance describing the open-phase of the vocal fold vibration). However few methods cope with estimation of high-frequency spectral tilt to describe the return-phase of the vocal fold vibration, which is crucial to the perception of vocal effort. This paper proposes an improved version of the well-known Iterative Adaptive Inverse Filtering (IAIF) called GFM-IAIF. GFM-IAIF includes a full spectral model of the glottis that incorporates both glottal formant and spectral tilt features. Comparisons with the standard IAIF method show that while GFM-IAIF maintains good performance on vocal tract removal, it significantly improves the perceptive timbral variations associated to vocal effort. * O. Perrotin and I.V. McLoughlin are with the
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