307 research outputs found
fofb~ and fog~ objects for Max/MSP software and accompanying demonstration/tutorial materials
No full textDetails:
The software is co-authored by Professor Clarke (principal researcher, 75% contribution) with Xavier Rodet (25%, IRCAM,
Paris). Together, the source code for the two objects (written in the C programming language) comprises over 2,500 lines of
code. The development of these objects and the underlying research are described in more detail in the fully refereed
conference paper in the portfolio. The objects are distributed by IRCAM internationally as part of the IRCAM Forum.
Context:
These algorithms are developed from Clarke’s own earlier FOF and FOG unit-generators for MIT’s Csound software package
and were inspired by IRCAM’s CHANT software developed by Xavier Rodet et al. The objects were developed to work in the
different environment of Max/MSP, in particular to take advantage of its orientation towards real-time work. This involved
researching methods for optimising code for real-time use and for adapting control parameters to facilitate real-time operation
and their creative use. Although the underlying algorithms remain in many respects unchanged, their implementation in this new
environment involved significant research and recoding. The fog~ object was also significantly extended to incorporate the
potential for use with data derived from Pitch Synchronous Overlap and Add (PSOLA) analyses of the sound being processed.
This approach provides ways of time stretching and pitch shifting sounds, substantially reducing the side effects that normally
arise with granular based approaches. The design permits other data from the analysis to be used to control the
voiced/unvoiced components of the sound, as well as vibrato and transients
The multiresolution Fourier transform and its application to polyphonic audio analysis
No full textMany people listen to, or at least hear, some form of music almost every day of their lives.
However, only some of the processes involved in creating the sensations and emotions evoked by
the music are understood in any detail. The problem of unravelling these processes has been much
less thoroughly investigated than the comparable topics of speech and image recognition; this has
almost certainly been caused by the existence of a greater number of applications awaiting this
knowledge. Nevertheless, the area of music perception has attracted some attention over the last
few decades and there is an increasing interest in the subject largely arising from the availability of
suitably powerful technology. It is becoming feasible to use such technology to construct artificial
hearing devices which attempt to reproduce the functionality of the human auditory system. The
construction of such devices is both a powerful method of verifying operational theories of the
human auditory system and may ultimately provide a means of analysing music in more detail
than man. In addition to the analytical benefits, techniques developed in this manner are readily
applicable to the creative aspects of music, such as the composition of new music and musical
sounds
One and Two Mass Model Oscillations for Voice and Instruments
No full text: Two movements seem possible for the lips of the trumpet player, upward and outward striking. A basic model, using one or the other movement, is proposed so that its behavior can more easily be understood and controlled. The two movements are compared. The model appears as a dynamic system of two loops coupled by a nonlinear function, one loop describes the dynamics of the single mass and the other represents the acoustic wave in the bore. Being much simpler, this model is simulated in real time on a workstation, allowing for easy experimentation of different designs and parameter values. I. Introduction Physical models consisting of an instantaneous nonlinearity and a linear feedback loop are relatively easy to understand and control [Rodet 93a]. But in the case of the trumpet or of the voice, the assumption of a massless reed, hence of an instantaneous nonlinearity, cannot be maintained. The movement of the lips of the trumpet player is discussed for instance in [Yoshikawa 95]. It s..
Physical Constraints for the Control of a Physical Model of a Trumpet: Wim D'haes, Dirk van Dyck and Xavier Rodet
No full textcote interne IRCAM: Dhaes02cNone / NoneNational audiencePhysical Constraints for the Control of a Physical Model of a Trumpe
The Diphone program: New features, new synthesis methods and experience of musical use
No full textGeneralized Diphone Control is a powerful means of building a musical phrase from dictionaries of analysed sound units by building sequences of units and concatenating and articulating them. We present new developments and features of the program Diphone 2.0, additive analysis ported on Macintosh, control of the Chant synthesis model and a Chant synthesis engine. We present also the experience that has been obtained about its musical usage after a year of work with musicians and composers. 1. Introduction Generalised diphone control and synthesis has first been proposed for musical applications in [Rodet 88] and an implementation has been described in [Depalle 93]. The first version (v. 0.9) of the Diphone program for Apple Macintosh has been presented at ICMC-96 in Hong-Kong and has been delivered in the IRCAM Forum. Version 2.0, which we present in this paper, will be delivered at the Forum in October 1997. Generalized Diphone control is a powerful means of building a musical phrase..
Control Parameter Estimation for a Physical Model of a Trumpet Using Pattern Recognition: Wim D'haes, Dirk van Dyck and Xavier Rodet
No full textcote interne IRCAM: Dhaes02dNone / NoneNational audienceControl Parameter Estimation for a Physical Model of a Trumpet Using Pattern Recognitio
Rodet: All-pole spectral envelope modelling with order selection for harmonic signals
No full textABSTRACT We present a study into all-pole spectral envelope estimation for the case of harmonic signals. We address the problem of the selection of the model order and propose to make use of the fact that the spectral envelope is sampled by means of the harmonic structure to derive a reasonable choice for an appropriate model order. The experimental investigation uses synthetic ARMA featured signals with varying fundamental frequency and differing model structure to evaluate the performance of the selected all-pole models. The experimental results confirm the relation between optimal model order and the fundamental frequency
Studies and Improvements in Automatic Classification of Musical Sound Samples
No full textcote interne IRCAM: Livshin03bNone / NoneNational audienceIn this article we shall deal with automatic classification of sound samples and ways to improve the classification results: We describe a classification process which produces high classification success percentage (over 95% for musical instruments) and compare the results of three classification algorithms: Multidimensional Gauss, KNN and LVQ. Next, we introduce several algorithms to improve the sound database self-consistency by removing outliers: LOO, IQR and MIQR. We present our efficient process for Gradual Elimination of Descriptors using Discriminant Analysis (GDE) which improves a previous descriptor selection algorithm (Peeters and Rodet 2002). It also enables us to reduce the computation complexity and space requirements of a sound classification process according to specific accuracy needs. Moreover, it allows finding the dominant separating characteristics of the sound samples in a database according to classification taxonomy. The article ends by showing that good classification results do not necessarily mean generalized recognition of the dominant sound source characteristics, but the classifier might actually be focused on the specific attributes of the classified database. By enriching the learning database with diverse samples from other databases we obtain a more general classifier. The dominant descriptors provided by GDE are then more closely related to what is supposed to be the distinctive characteristics of the sound sources
Rényi information measures for spectral change detection
No full textChange detection within an audio stream is an important task in several domains, such as classification and segmentation of a sound or of a music piece, as well as indexing of broadcast news or surveillance applications. In this paper we propose two novel methods for spectral change detection without any assumption about the input sound: they are both based on the evaluation of information measures applied to a time-frequency representation of the signal, and in particular to the spectrogram. The class of measures we consider, the Rényi entropies, are obtained by extending the Shannon entropy definition: a biasing of the spectrogram coefficients is realized through the dependence of such measures on a parameter, which allows refined results compared to those obtained with standard divergences. These methods provide a low computational cost and are well-suited as a support for higher level analysis, segmentation and classification algorithms
Audio Engineering Society Convention Paper Presented at the 120th Convention
No full textThis convention paper has been reproduced from the author's advance manuscript, without editing, corrections, or consideration by the Review Board. The AES takes no responsibility for the contents. Additional papers may be obtained by sending reques
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