Sergio Canazza

An Abstract Control Space for Communication of Sensory Expressive Intentions in Music Performance

Expressiveness is not an extravagance: instead, expressiveness plays a critical role in rational decision-making, in perception, in human interaction, in human emotions and in human intelligence. These facts, combined with the development of new informatics systems able to recognize and understand different kinds of signals, open new areas for research. A new model is suggested for computer understanding of sensory expressive intentions of a human performer and both theoretical and practical applications are described for human-computer interaction, perceptual information retrieval, creative arts and entertainment. Recent studies demonstrated that by opportunely modifying systematic deviations introduced by the musician it is possible to convey different sensitive contents, such as expressive intentions and/or emotions. We present an space, that can be used as a user interface. It represents, at an abstract level, the expressive content and the interaction between the performer and an expressive synthesizer.

Modeling and control of expressiveness in music performance

Expression is an important aspect of music performance. It is the added value of a performance and is part of the reason that music is interesting to listen to and sounds alive. Understanding and modeling expressive content communication is important for many engineering applications in information technology. For example, in multimedia products, textual information is enriched by means of graphical and audio objects. In this paper, we present an original approach to modify the expressive content of a performance in a gradual way, both at the symbolic and signal levels. To this purpose, we discuss a model that applies a smooth morphing among performances with different expressive content, adapting the audio expressive character to the users desires. Morphing can be realized with a wide range of graduality (from abrupt to very smooth), allowing adaptation of the system to different situations. The sound rendering is obtained by interfacing the expressiveness model with a dedicated postprocessing environment, which allows for the transformation of the event cues. The processing is based on the organized control of basic audio effects. Among the basic effects used, an original method for the spectral processing of audio is introduced.

Audio Morphing Different Expressive Intentions for Multimedia Systems

Web Extras: Sample audio files and view a demo of the audio authoring tool.Download Real Jukebox for listening to the mp3 filesSonatina in sol (by Beethoven) played neutral (without any expressive intentions)Expressive performance of Sonatina in sol generated by the model in a symbolic way (that is, as a MIDI file)Sonata K545 (by Mozart) played neutral (without any expressive intentions)Expressive performance of Sonata K545 generated by the model in a symbolic way (that is, as a MIDI file)Expressive performance of Sonata in A Major Op. V(by Corelli) generated by the audio authoring tool (using the audio postprocessing tool)

Restoration of Audio Documents by Means of Extended Kalman Filter

We present some results on audio restoration obtained with an algorithm that solves the problems of broadband noise filtering, signal parameters tracking, and impulsive noise removal by using the Extended Kalman Filter (EKF) theory. We show that, to achieve maximum performance, it is essential to optimize the EKF implementation. To this purpose, having to cope with the nonstationarity of the audio signal, we use two properly combined EKF filters (forward and backward), and introduce a bootstrapping procedure for model tracking. The careful combination of the proposed techniques and an accurate choice of some critical parameters, allows to improve the performance of the EKF algorithm. The presented procedure is validated by listening tests.

Sonological Analysis of Clarinet Expressivity

Seven different performances of a fragment of W. A. Mozarts Concert for Clarinet (K622) were recorded. Each performance was inspired giving the player a sensorial type adjective in order to suggest various kinds of expressive intentions. Certain particularly significant sonological parameters were then measured. These were linked to the time factor (correlated to variations in the amplitude and duration) and frequency (variations in the timbre). The principle aim was to identify which physical parameters, and how many of them, were subject to modifications when the expressive intention of the performer was varied. The sonological analysis shows how a performer changes the physical parameters of frequency and time in function of expressive performance of a score. The evaluation of a synthesized performance, based of the values of the parameters measured, confirms the validity of the procedure.

Adaptive Time Delay Estimation Using Filter Length Constraints for Source Localization in Reverberant Acoustic Environments

Adaptive time delay estimation based on blind system identification (BSI) focuses on the impulse responses between a source and a microphone to estimate the time difference of arrival (TDOA) in reverberant environments. In this letter, we consider the adaptive eigenvalue decomposition (AED) BSI method based on the normalized multichannel frequency-domain least mean square (NMCFLMS) algorithm. We show that the use of filter length constraints (FLC) based on the maximum TDOA between microphones improves the performance of the NMCFLMS filter for the localization of different sound types in highly reverberant environments. The experimental results demonstrate the improvement of the proposed method for reverberation times (RT60) of up to 2 s. Applications for this method include teleconferencing systems, musical interfaces, videogames, and monitoring systems.

A systemic approach to the preservation of audio documents: methodology and software tools

This paper presents amethodology for the preservation of audio documents, the operational protocol that acts as the methodology, and an original open source software system that supports and automatizes several tasks along the process. The methodology is presented in the light of the ethical debate that has been challenging the international archival community for the last thirty years. The operational protocol reflects the methodological principles adopted by the authors, and its effectiveness is based on the results obtained in recent research projects involving some of the finest audio archives in Europe. Some recommendations are given for the rerecording process, aimed at minimizing the information loss and at quantifying the unintentional alterations introduced by the technical equipment. Finally, the paper introduces an original software systemthat guides and supports the preservation staff along the process, reducing the processing timing, automatizing tasks, minimizing errors, and using information hiding strategies to ease the cognitive load. Currently the software system is in use in several international archives.

Entertaining listening by means of the Stanza Logo-Motoria: an Interactive Multimodal Environment

Abstract This article presents an Interactive Multimodal Environment (IME), the Stanza Logo-Motoria, designed to support learning in primary schools. In particular we describe the use of this system as a tool (a) to practice listening to English as a Second Language (ESL) and (b) to enable children with severe disabilities to perform an interactive listening. We document the ongoing experimentation of the Stanza Logo-Motoria in ESL lessons and report its encouraging results. Moreover, we explain how it may be possible, by means of the Stanza Logo-Motoria, to redesign traditional learning environments in order to allow pupils to experience listening as an active and engaging experience.

Methodologies and tools for audio digital archives

In response to the proposal of digitizing the entire back-run of several European audio archives, many research projects have been carried out in order to discover the technical issues involved in making prestigious audio documents digitally available, which are related to the A/D transfer process and supervised metadata extraction. This article gives an innovative approach to metadata extraction from such a complex source material. This article also describes the protocols defined, the processes undertaken, the results ascertained from several audio documents preservation projects and the techniques used. In addition, a number of recommendations are given for the re-recording process, aimed at minimizing the information loss and to automatically measure the unintentional alterations introduced by the A/D equipment.

Toward a Methodology for the Restoration of Electroacoustic Music

In recent years the evolution of digital technologies for the coding of audio signals has lead to new perspectives in the field of preservation and fruition of audio archives, even though new models for digital audio archives have not been yet developed. The most important international archives make use today of digital databases, and an increasing number of digital supports (DAT, CD-A, DVD-A). At the same time, in the field of musical philology, restored editions of electroacustic operas have never been released: these would lead to scientific problems never addressed so far, about methodologies and algorithms. We aim at applying restoration algorithms to recordings of electronic music; this repertoire has some peculiar problems: it requires a philological analysis of the piece, as many of the synthesized sounds have acoustic properties in common with noise, both impulsive and white. This work presents the results of an experiment aimed to evaluate the quality of different audio restoration algorithms based on different methods, based both time and frequency domain.