The collision interaction mechanism between a vibrating string and a non-resonant obstacle is at the heart of many musical instruments. This paper focuses on the identification of perceptually salient auditory features related to this phenomenon. The objective is to design a signal-based synthesis process, with an eye towards developing intuitive control strategies. To this end, a database of synthesized sounds is assembled through physics-based emulation of a string/obstacle collision, in order to characterize the effect of collisions on time-frequency content. The investigation of this database reveals characteristic time-frequency patterns related to the position of the obstacle during the interaction. In particular, a frequency shift of certain modes is apparent for strong interactions, which, alongside the generation of new frequency components, leads to increased perceived roughness and inharmonicity. These observations enable the design of a real-time compatible signal-based sound synthesis process, with a mapping of synthesis parameters linked to the perceived location of the obstacle. The accuracy of the signal model with respect to the physical model sound output and recorded sounds was evaluated through listening tests: time-frequency patterns reproduced by the signal model enabled listeners to precisely recognize the transverse location of the obstacle.

Most recent vehicles are equipped with touchscreens, which replace arrays of buttons that control secondary driving functions, such as temperature level, strength of ventilation, GPS, or choice of radio stations. While driving, manipulating such interfaces can be problematic in terms of safety, because they require the drivers’ sight. In this article, we develop an innovative interface, MovEcho, which is piloted with gestures and associated with sounds that are used as informational feedback. We compare this interface to a touchscreen in a perceptual experiment that took place in a driving simulator. The results show that MovEcho allows for a better visual task completion related to traffic and is preferred by the participants. These promising results in a simulator condition have to be confirmed in future studies, in a real vehicle with a comparable expertise for both interfaces.

This paper aims to reveal the efficiency of sonification strategies in terms of rapidity, precision, and overshooting in the case of a one-dimensional guidance task. The sonification strategies are based on the four main perceptual attributes of a sound (pitch, loudness, duration/tempo, and timbre) and classified with respect to the presence or not of one or several auditory references. Perceptual evaluations are used to display the strategies in a precision/rapidity space and enable prediction of user behavior for a chosen sonification strategy. The evaluation of sonification strategies constitutes a first step toward general guidelines for sound design in interactive multimedia systems that involve guidance issues.

This paper presents a physically inspired source-filter model for rolling sound synthesis. The model, which is suitable for real-time implementation, is based on qualitative and quantitative observations obtained from a physics-based model described in the literature. In the first part of the paper, the physics-based model is presented, followed by a perceptual experiment, whose aim is to identify the perceptually relevant information characterizing the rolling interaction. On the basis of this experiment, we hypothesize that the particular pattern of the interaction force is responsible for the perception of a rolling object. A complete analysis-synthesis scheme of this interaction force is then provided, along with a description of the calibration of the proposed source-filter sound synthesis process. Finally, a mapping strategy for intuitive control of the proposed synthesis process (i.e. size and velocity of the rolling object and roughness of the surface) is proposed and validated by a listening test.

This article addresses the question of synthesis and control of sound attributes from a perceptual point of view. We focused on an attribute related to the general concept of motion evoked by sounds. To investigate this concept, we tested 40 monophonic abstract sounds on listeners via a questionnaire and drawings, using a parametrized custom interface. This original procedure, which was defined with synthesis and control perspectives in mind, provides an alternative means of determining intuitive control parameters for synthesizing sounds evoking motion. Results showed that three main shape categories (linear, with regular oscillations, and with circular oscillations) and three types of direction (rising, descending, and horizontal) were distinguished by the listeners. In addition, the subjects were able to perceive the low-frequency oscillations (below 8 Hz) quite accurately. Three size categories (small, medium, and large) and three levels of randomness (none, low amplitude irregularities, and high amplitude irregularities) and speed (constant speed and speeds showing medium and large variations) were also observed in our analyses of the participants' drawings. We further performed a perceptual test to confirm the relevance of the contribution of some variables with synthesized sounds combined with visual trajectories. Based on these results, a general typology of evoked motion was drawn up and an intuitive control strategy was designed, based on a symbolic representation of continuous trajectories (provided by devices such as motion capture systems, pen tablets, etc.). These generic tools could be used in a wide range of applications such as sound design, virtual reality, sonification, and music.

We propose here a new approach together with a corresponding class of algorithms for offline estimation of linear operators mapping input to output signals. The operators are modeled as multipliers, i.e., linear and diagonal operator in a frame or Bessel representation of signals (like Gabor, wavelets ...) and characterized by a transfer function. The estimation problem is formulated as a regularized inverse problem, and solved using iterative algorithms, based on gradient descent schemes. Various estimation problems, which differ by a choice for the regularization function, are studied in the case of Gabor multipliers. The transfer function actually provides a meaningful interpretation of the differences between the two signals or signal classes under consideration, and examples are discussed. Furthermore, examples of signal transformations with such Gabor transfer functions are also given.

The inverse fast Fourier transform (IFFT) method is a time-frequency technique which was proposed to alleviate the complexity of the additive sound synthesis method in real-time applications. However, its application is limited by its inherent tradeoff between time and frequency resolutions, which are determined by the number of frequencies used for time-frequency processing. In a previous work, the authors proposed a frequency-refining technique for overcoming this frequency limitation, permitting achieving any time and frequency resolution using a small number of frequencies. In this correspondence we extend this work, by proposing a time-refining technique which permits overcoming the time resolution limitation for a given number of frequencies. Additionally, we propose an alternative to the frequency-refining technique proposed in our previous work, which requires about half the computations. The combination of these two results permits achieving any time and frequency resolution for any given number of frequencies. Using this property, we find the number of frequencies which minimizes the overall complexity. We do so considering two different application scenarios (i.e., offline sound design and online real-time synthesis). This results in a major complexity reduction in comparison with the design proposed in our previous work.

In this paper, we focused on the identification of the perceptual properties of impacted materials to provide an intuitive control of an impact sound synthesizer. To investigate such properties, impact sounds from everyday life objects, made of different materials (wood, metal and glass), were recorded and analyzed. These sounds were synthesized using an analysis–synthesis technique and tuned to the same chroma. Sound continua were created to simulate progressive transitions between materials. Sounds from these continua were then used in a categorization experiment to determine sound categories representative of each material (called typical sounds). We also examined changes in electrical brain activity (using event related potentials (ERPs) method) associated with the categorization of these typical sounds. Moreover, acoustic analysis was conducted to investigate the relevance of acoustic descriptors known to be relevant for both timbre perception and material identification. Both acoustic and electrophysiological data confirmed the importance of damping and highlighted the relevance of spectral content for material perception. Based on these findings, controls for damping and spectral shaping were tested in synthesis applications. A global control strategy, with a three-layer architecture, was proposed for the synthesizer allowing the user to intuitively navigate in a “material space” and defining impact sounds directly from the material label. A formal perceptual evaluation was finally conducted to validate the proposed control strategy.

The inverse fast Fourier transform (FFT) method was proposed to alleviate the computational complexity of the additive sound synthesis method in real-time applications, and consists in synthesizing overlapping blocks of samples in the frequency domain. However, its application is limited by its inherent tradeoff between time and frequency resolution. In this paper, we propose an alternative to the inverse FFT method for synthesizing colored noise. The proposed approach uses subband signal processing to generate time-frequency noise with an autocorrelation function such that the noise obtained after converting it to time domain has the desired power spectral density. We show that the inverse FFT method can be interpreted as a particular case of the proposed method, and therefore, the latter offers some extra design flexibility. Exploiting this property, we present experimental results showing that the proposed method can offer a better tradeoff between time and frequency resolution, at the expense of some extra computations.

Nowadays, interactive 3-D environments tend to include both synthesis and spatialization processes to increase the realism of virtual scenes. In typical systems, audio generation is created in two stages: first, a monophonic sound is synthesized (generation of the intrinsic timbre properties) and then it is spatialized (positioned in its environment). In this paper, we present the design of a 3-D immersive synthesizer dedicated to environmental sounds, and intended to be used in the framework of interactive virtual reality applications. The system is based on a physical categorization of environmental sounds (vibrating solids, liquids, aerodynamics). The synthesis engine has a novel architecture combining an additive synthesis model and 3-D audio modules at the prime level of sound generation. An original approach exploiting the synthesis capabilities for simulating the spatial extension of sound sources is also presented. The subjective results, evaluated with a formal listening test, are discussed. Finally, new control strategies based on a global manipulation of timbre and spatial attributes of sound sources are introduced.

This paper presents a sound synthesis model that reproduces impact sounds by taking into account both the perceptual and the physical aspects of the sound. For that, we used a subtractive method based on dynamic filtering of noisy input signals that simulates the damping of spectral components. The resulting sound contains the perceptual characteristics of an impact on a given material. Further, the addition of very few modal contributions-using additive or banded digital waveguide synthesis-together with a bandpass filtering taking into account the interaction with the excitator, allows realistic impact sounds to be synthesized. The synthesis parameters can be linked to a perceptual notion of material and geometry of the sounding object. To determine the synthesis parameters, we further address the problem of analysis-synthesis aiming at reconstructing a given impact sound. The physical parameters are extracted through a time-scale analysis of natural sounds. Examples are presented for sounds generated by impacting plates made of different materials and a piano soundboard.

We present a novel way of implementing an audio coder aimed at good sound quality for very low bit-rates. The signal is first decomposed into a tonal part, transients and a stochastic part. The corresponding coefficients are quantized according to a psychoacoustic model, encoded with arithmetic coding and bit-packed. Some perceptually important high-level psychoacoustic attributes are also encoded, and used by the decoder in a post-processing unit.

One important field in the framework of computer music concerns the modeling of sounds. In order to design digital models mirroring as closely as possible a real sound and permitting in addition intimate transformations by altering the synthesis parameters, we look for a signal model based on additive synthesis whose parameters are estimated by the analysis of real sounds. This model is relevant from both the physical and perceptual points of view, especially when the sound to be analyzed comes from a musical instrument. We present some techniques, mostly unpublished, based on time-frequency representations which make possible the estimation of relevant parameters such as frequency and amplitude modulation laws corresponding to each spectral component of the sound. The techniques described extend the results presented by Delprat et al. (see IEEE Trans. Inform. Theory, vol.38, p.644-65, March 1992). These methods are then transposed to broadband signals, allowing the characterization of transients.

The behavior of the continuous wavelet and Gabor coefficients in the asymptotic limit using stationary phase approximations are investigated. In particular, it is shown how, under some additional assumptions, these coefficients allow the extraction of some characteristics of the analyzed signal, such as frequency and amplitude modulation laws. Applications to spectral line estimations and matched filtering are briefly discussed.>

This paper starts with a brief discussion of so-called wavelet transforms, i.e., decompositions of arbitrary signals into localized contributions labelled by a scale parameter. The main features of the method are first illustrated through simple mathematical examples. Then we present the first applications of the method to the recognition and visualisation of characteristic features of speech and of musical sounds.