Download Measuring Diffusion in a 2D Digital Waveguide Mesh The digital waveguide mesh is a method by which the propagation of sound waves in an acoustic system can be simulated. An important consideration in modelling such systems is the accurate modelling of reflection characteristics at boundaries and surfaces. A significant property of an acoustic boundary is its diffusivity. In fact partially diffuse sound reflections are observed at most real acoustic surfaces and so this is an important consideration when implementing a digital waveguide mesh model. This paper presents a method for modelling diffusion that offers a high degree of control. The model is implemented with varying amounts of diffusivity, and a method for measuring its diffusive properties is outlined. Results for the model are presented and a method to calculate the diffusion coefficient is described.
Download Performing Expressive Rhythms with BillaBoop Voice-Driven Drum Generator In a previous work we presented a system for transcribing spoken rhythms into a symbolic score. Thereafter, the system was extended to process the vocal stream in real-time in order to allow a musician to use it as a voice-driven drum generator. Extensions to this work are the following. First we achieved a study of the system classification accuracy based on typical onomatopoeia used in western beat boxing, with the perspective of building a general supervised model for immediate use. Also, we want the user to be able to generate expressive rhythms, beyond the symbolic drum representation. Thus we considered a class-specific mapping of continuous vocal stream descriptors with either effects or synthesis parameters of the drum generator. The extraction of the symbolic drum stream is implemented in the BillaBoop VST Core plug-in. The class-specific mapping and the sound synthesis are carried out in Plogue Bidule 1 framework. All these components are integrated into a low-latency application that allows its use for live performances.
Download Extraction of long-term structures in musical signals using the empirical mode decomposition Long-term musical structures provide information concerning rhythm, melody and the composition. Although highly musically relevant, these structures are difficult to determine using standard signal processing. In this paper, a new technique based on the time-domain empirical mode decomposition is explained which enables us to analyse both short-term information and long-term structures in musical signals. It provides insight into perceived rhythms and their relationship to the signal. The technique is explained, and results are reported and discussed. Keywords: Empirical Mode Decomposition (EMD), Music Analysis, Santur, Long-term Structures, Fundamental Frequency, Rhythm.
Download An Eyes-free User Interface Controlled by Finger Snaps A novel way of controlling a simple user interface based on detecting and localizing finger snaps of the user is presented. The analysis method uses binaural signals recorded from the ears of the user. Transient sounds are first detected from a continuous audio stream, followed by cross-correlation based localization and simple band-energy ratio based classification. The azimuth plane around the user is divided into three sectors, each of which corresponds to one of the three “buttons” in the interface. As an example, the interface is applied for controlling the playlist of an MP3 player. The algorithm performance was evaluated using a real-world recording. While the algorithm looks promising, more research is needed before it is ready for commercial applications.
Download Analysis-assisted sound processing with audiosculp Digital audio effects using phase vocoder techniques are currently in widespread use. However, their interfaces often hide vital parameters from the user. This fact, and the generally limited ways in which sound designing and compositional tools can represent sounds and their spectral content, complicates the effective use of the full potential of modern effect processing algorithms. This article talks about ways in which to use analysis to obtain better processing results with phase vocoder based effects, and how these techniques are implemented in IRCAM's AudioSculpt application. Also discussed are the advantages of using the SDIF format to exchange analysis data between various software components, which facilitates the integration of new analysis and processing algorithms.
Download REAL-TIME SIGNAL PROCESSING SYSTEM PROPOSAL AND IMPLEMENTATION This paper intends to describe the desirable features of a complete, powerful and highly customizable real-time audio algorithm implementation system, and to provide the guidelines to its implementation. The goal is to design a platform by means of which new sophisticated audio algorithms can be developed, tested and used with the minimum effort. The idea is to build large complex processing systems based on elemental building blocks which may interact in any possible manner. This way, by connecting existing, proved modules, such as filters, noise gates, or any new specific module, complex processes can be achieved and tested in a realtime environment with the minimum possible effort. The buildingblock philosophy would also make such a system very suitable for educational purposes, as it would make possible to ‘hear’ in realtime a particular complex algorithm with and without one of its blocks (a filter, for example), thus showing its importance.
Download Scalable Perceptual Mixing and Filtering of Audio Signals Using an Augmented Spectral Representation Many interactive applications, such as video games, require processing a large number of sound signals in real-time. This paper proposes a novel perceptually-based and scalable approach for efficiently filtering and mixing a large number of audio signals. Key to its efficiency is a pre-computed Fourier frequency-domain representation augmented with additional descriptors. The descriptors can be used during the real-time processing to estimate which signals are not going to contribute to the final mixture. Besides, we also propose an importance sampling strategy allowing to tune the processing load relative to the quality of the output. We demonstrate our approach for a variety of applications including equalization and mixing, reverberation processing and spatialization. It can also be used to optimize audio data streaming or decompression. By reducing the number of operations and limiting bus traffic, our approach yields a 3 to 15-fold improvement in overall processing rate compared to brute-force techniques, with minimal degradation of the output.
Download An auditory 3D file manager designed from interaction patterns This paper shows the design, implementation and evaluation of an auditory user interface for a file-manager application. The intention for building this prototype was to prove concepts developed to support user interface designers with design patterns in order to create robust and efficient auditory displays. The paper describes the motivation for introducing a mode-independent meta domain in which the design patterns were defined to overcome the problem of translating mainly visual concepts to the auditory domain. The prototype was implemented using the IEM Ambisonics libraries for Pure Data to produce high quality binaural audio rendering and used headtracking and a joystick as the main interaction devices.
Download Conjugate gradient techniques for multichannel acoustic echo cancellation Conjugate Gradient (CG) techniques are suitable for resolution of time-variant system identification problems: adaptive equalization, echo cancellation, active noise cancellation, linear prediction, etc. These systems can be seen as optimization problems and CG techniques can be used to solve them. It has been demonstrated that, in the single-channel case, the conjugate gradient techniques provide a similar solution in terms of convergence rate than those provided by the recursive least square (RLS) method, involving higher complexity than the least mean square (LMS) but lower than RLS without stability issues. The advantages of these techniques are especially valuable in the case of high complexity and magnitude problems like multi-channel systems. This work develops CG algorithm for the adaptive MIMO (multiple-input and multiple-output) systems and tests it by solving a multichannel acoustic echo cancellation (MAEC) problem.