Download Filters, Delays, Modulations and Demodulations: A tutorial
A set of basic signal processing building blocks is presented and the implementation issues, that are relevant to musical applications, are underlined. in this text a filter classification is first outlined then implementation schemes for analog and digital lowpass-filters, within the canonical and the state-variable structure, are given. As well as higher order filters according to the Butterworth design are suggested, delay-based effects such as comb-filter, resonator, flanger, chorus, slapback and echo are reviewed. Phasing is mentioned and normalization schemes for some filters are proposed. Modulators (ring, amplitude, single side band) as well as demodulators (AM-detector, amplitude follower with independent attack and release times, instantaneous envelope follower) are reviewed and their applications listed.
Download Embedded Audio Compression using Wavelets and improved Psychoacoustic Models
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Download Dispersive and Pitch-Synchronous Processing of Sounds
The aim of this paper is to present results on digital processing of sounds by means of both dispersive delay lines and pitch-synchronous transforms in a unified framework. The background on frequency warping is detailed and applications of this technique are pointed out with reference to the existing literature. These include transient extraction, pitch shifting, harmonic detuning and auditory modeling.
Download Numerical Model of the Sound Field inside Cars for the Creation of Virtual audible Reconstructions
The research undertaken aims to the development of a new numerical predictive technique, capable of producing such a detailed simulation, so that it can be used for the creation of “virtual” sound samples, theoretically indistinguishable from binaural recordings inside a real car. This way, a proposed (not existing yet) sound system could be compared with the existing ones by direct listening tests. This paper describes the implementation of the numerical model, based on Boundary Elements for low frequencies and Pyramid Tracing for higher frequencies. The results of the two models are merged together, producing a binaural impulse response which can be convolved with the music signal. The performances of the system were evaluated by listening tests, in comparison with experimental measurements taken in a 1:5 scale model of a car compartment.
Download Inverse numerical filters for linearisation of loudspeaker's
In this paper a non-linear loudspeaker model, which accurately reproduces the low frequency behavior, is presented. This description, derived from an extension of the well known Small-Thiele equations, requires far less computational time and memory space than generic non linear structures. Moreover a noticeable further reduction of the number of operations and of the memory cells required has been achieved by means of a multirate architecture. Inversion of the proposed model allows digital prefiltering of the electrical signal in order to compensate for the nonidealities of the electro-acoustic conversion. The above filter structure implemented on a digital signal processor, placed between the audio signal source and the power amplifier allows effective compensation of loudspeaker linear (both magnitude and phase) and non-linear distortion. Measurement results obtained with a commercial woofer are discussed
Download Aneto: a tool for prosody analysis of speech
The developed tool provides utilities for prosody analysis and labeling of voice signals. It works under Windows 95 and Windows NT environments and uses the Microsoft Win32 application programming interface (API) for audio playing and recording. The application detects the prosody of speech signal and then the original intonation can be stylized in order to observe the pitch contour. Besides, the original intonation can be easily modified and it is possible to resynthesize the voice signal according to the new intonation. Listening to the resynthesized signal, the user can evaluate the results of the prosodic modification.
Download Enhanced quality and variety for chorus/flange units
Chorus and flanger are two classical effects that can be found in any current digital audio FX unit. Available chorus/flanger just translate into digital implementations the behaviour and parameters typical of cost-sensitive analogue realizations. But they can be greatly improved on a digital implementation with simple additions to the software. After an introduction to chorus/flange and their digital realizations, we address two such additions (randomized modulation law, and application of energy envelope for noise control), and then discuss further improvements easily achiveable by digital means (multiscale implementations).
Download Sound Generation with Aperiodic Ordered Systems
Aperiodic 1D systems introduced in physics in the field of quasicrystals are used in this work, in order to generate selfsimilar aperiodic time structures. The Fourier analysis of series of impulses distributed in time in a non periodic but ordered way, shows that for some cases the spectrum has a discrete part that can be used for sound synthesis.
Download Sound and Music transformation environments : a 20 years experiment at the french Groupe de Recherches Musicales
Since the 1970s, the Inaís Groupe de Recherches Musicales (Music Research Group) has been developing the computer tools needed for the creation of electroacoustic works by the composers it invites. The history of the successive projects and the conclusions that can be drawn, offer a wealth of information concerning the experience acquired by the researchers in contact with the musicians.