Download Recent developments in PWSYNTH PWSynth was originally a visual synthesis language situated in PatchWork. Recently our research team has started a complete rewrite of the system so that it can be adapted to our new programming environment called PWGL. In this paper we present the main differences of the old and new systems. These include switching from C to C++, efficiency issues, interface between PWGL and the synthesis engine, and a novel copy-synth-patch scheme.
Download Extracting automatically the perceived intensity of music titles We address the issue of extracting automatically high-level musical descriptors out of their raw audio signal. This work focuses on the extraction of the perceived intensity of music titles, that evaluates how energic the music is perceived by listeners. We present here first the perceptive tests that we have conducted, in order to evaluate the relevance and the universality of the perceived intensity descriptor. Then we present several methods used to extract relevant features used to build automatic intensity extractors: usual Mpeg7 low level features, empirical method, and features automatically found using our Extractor Discovery System (EDS), and compare the final performances of their extractors.
Download The wave digital reed: A passive formulation In this short paper, we address the numerical simulation of the single reed excitation mechanism. In particular, we discuss a formalism for approaching the lumped nonlinearity inherent in such a model using a circuit model and the application of wave digital filters (WDFs), which are of interest in that they allow simple stability verification, a property which is not generally guaranteed if one employs straightforward numerical methods. We present first a standard reed model, then its circuit representation, then finally the associated wave digital network. We then enter into some implementation issues, such as the solution of nonlinear algebraic equations, and the removal of delay-free loops, and present simulation results.
Download Analysis and resynthesis of quasi-harmonic sounds: an iterative filterbank approach We employ a hybrid state-space sinusoidal model for general use in analysis-synthesis based audio transformations. This model, which has appeared previously in altered forms (e.g. [5], [8], perhaps others) combines the advantages of a source-filter model with the flexible, time-frequency based transformations of the sinusoidal model. For this paper, we specialize the parameter identification task to a class of “quasi-harmonic” sounds. The latter represent a variety of acoustic sources in which multiple, closely spaced modes cluster about principal harmonics loosely following a harmonic structure (some inharmonicity is allowed.) To estimate the sinusoidal parameters, an iterative filterbank splits the signal into subbands, one per principal harmonic. Each filter is optimally designed by a linear programming approach to be concave in the passband, monotonic in transition regions, and to specifically null out sinusoids in other subband regions. Within each subband, the constant frequencies and exponential decay rates of each mode are estimated by a Steiglitz-McBride approach, then time-varying amplitudes and phases are tracked by a Kalman filter. The instantaneous phase estimate is used to derive an average instantaneous frequency estimate; the latter averaged over all modes in the subband region updates the filter’s center frequency for the next iteration. In this way, the filterbank structure progressively adapts to the specific inharmonicity structure of the source recording. Analysissynthesis applications are demonstrated with standard (time/pitchscaling) transformation protocols, as well as some possibly novel effects facilitated by the “source-filter” aspect.
Download System analysis and performance tuning for broadcast audio fingerprinting An audio fingerprint is a content-based compact signature that summarizes an audio recording. Audio Fingerprinting technologies have recently attracted attention since they allow the monitoring of audio independently of its format and without the need of meta-data or watermark embedding. These technologies need to face channel robustness as well as system accuracy and scalability to succeed on real audio broadcasting environments. This paper presents a complete audio fingerprinting system for audio broadcasting monitoring that satisfies the above system requirements. The system performance is enhanced with four proposals that required detailed analysis of the system blocks as well as extense system tuning experiments.
Download Room simulation for binaural sound reproduction using measured spatiotemporal impulse responses In binaural sound reproduction systems the incorporation of room simulation is important to improve sound source localisation capabilities. Thus, the localisation error can be decreased, while equivalently an enhanced externality (out of head localisation) is achieved. Previously proposed works are based on simple geometrical approaches for room simulation. In this paper an alternative method using measured room impulse responses (RIRs) is presented. Therefore, it is possible to obtain a convincing acoustical image of an existing room. The RIRs are measured using a circular microphone array to capture both temporal and spatial information of the desired room.
Download A hierarchical approach to automatic musical genre classification A system for the automatic classification of audio signals according to audio category is presented. The signals are recognized as speech, background noise and one of 13 musical genres. A large number of audio features are evaluated for their suitability in such a classification task, including well-known physical and perceptual features, audio descriptors defined in the MPEG-7 standard, as well as new features proposed in this work. These are selected with regard to their ability to distinguish between a given set of audio types and to their robustness to noise and bandwidth changes. In contrast to previous systems, the feature selection and the classification process itself are carried out in a hierarchical way. This is motivated by the numerous advantages of such a tree-like structure, which include easy expansion capabilities, flexibility in the design of genre-dependent features and the ability to reduce the probability of costly errors. The resulting application is evaluated with respect to classification accuracy and computational costs.
Download A real-time audio rendering system for the Internet (iARS), embedded in an electronic music library (IAEM) The internet Audio Rendering System (iARS) is an Internet browser extension extending the browser’s capabilities with real-time signal processing. The proposed system allows to receive audio streams from the Internet and apply various audio algorithms with no additional computational power needed from the server. iARS is part of the Internet Archive of Electronic Music (IAEM) project which is also presented in this paper.The IAEM is intended to be a platform to access an extensive and distributed archive of electronic music. It combines collaborative tools, real time signal processing on the client side and the content of the archive to a powerful teaching, research and publishing tool.
Download Interpolation of long gaps in audio signals using the warped Burg's method This paper addresses the reconstruction of missing samples in audio signals via model-based interpolation schemes. We demonstrate through examples that employing a frequency-warped version of Burg’s method is advantageous for interpolation of long duration signal gaps. Our experiments show that using frequencywarping to focus modeling on low frequencies allows reducing the order of the autoregressive models without degrading the quality of the reconstructed signal. Thus a better balance between qualitative performance and computational complexity can be achieved.
Download The caterpillar system for data-driven concatenative sound synthesis Concatenative data-driven synthesis methods are gaining more interest for musical sound synthesis and effects. They are based on a large database of sounds and a unit selection algorithm which finds the units that match best a given sequence of target units. We describe related work and our C ATERPILLAR synthesis system, focusing on recent new developments: the advantages of the addition of a relational SQL database, work on segmentation by alignment, the reformulation and extension of the unit selection algorithm using a constraint resolution approach, and new applications for musical and speech synthesis.