Paper Archive

Raster scanning is a technique for generating or recording a video image by means of a line-by-line sweep, tantamount to a data mapping scheme between one and two dimensional spaces. While this geometric structure has been widely used on many data transmission and storage systems as well as most video displaying and capturing devices, its application to audio related research or art is rare. In this paper, a data mapping mechanism of raster scanning is proposed as a framework for both image sonification and sound visualization. This mechanism is simple, and produces compelling results when used for sonifying image texture and visualizing sound timbre. In addition to its potential as a cross modal representation, its complementary and analogous property can be applied sequentially to create a chain of sonifications and visualizations using digital filters, thus suggesting a useful creative method of audio processing. Special attention is paid to the rastrogram - raster visualization of sound - as an intuitive visual interface to audio data. In addition to being an efficient means of sound representation that provides meaningful display of significant auditory features, the rastrogram is applied to the area of sound analysis by visualizing characteristics of loop filters used for a Karplus-Strong model. A new sound synthesis method based on texture analysis/synthesis of the rastrogram is also suggested.

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This article discusses some developments relating to environments for Csound programming, composition and performance. It introduces the Csound 5 API and discusses its use in the development of a TclTk scripting interface, TclCsound. The three components of TclCsound are presented and discussed. A number of applications, from simple transport control of Csound to client-server networking are explained in some detail. The new multi-platform version of CECILIA is presented. Cecilia is the first Csound frontend to use the functionalities of TclCsound.

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A sound scene can be defined as any “environmental” sound that has a consistent background texture, with one or more potentially recurring foreground events. We describe a data-driven framework for analyzing, transforming, and synthesizing high-quality sound scenes, with flexible control over the components of the synthesized sound. Given one or more sound scenes, we provide well-defined means to: (1) identify points of interest in the sound and extract them into reusable templates, (2) transform sound components independently of the background or other events, (3) continually re-synthesize the background texture in a perceptually convincing manner, and (4) controllably place event templates over the background, varying key parameters such as density, periodicity, relative loudness, and spatial positioning. Contributions include: techniques and paradigms for template selection and extraction, independent sound transformation and flexible re-synthesis; extensions to a wavelet-based background analysis/synthesis; and user interfaces to facilitate the various phases. Given this framework, it is possible to completely transform an existing sound scene, dynamically generate sound scenes of unlimited length, and construct new sound scenes by combining elements from different sound scenes. URL: http://taps.cs.princeton.edu/

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This article introduces some new possibilities of audio manipulations and sound processing in the Computer-Aided Composition environment OpenMusic. Interfaces with underlying sound processing systems are described, with an emphasis on the use of the symbolic and visual programming environment for the design of sound computation processes.

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The phase vocoder is a standard frequency domain time-scaling technique suitable for polyphonic audio, but it generates annoying artifacts called phasiness, or loss of presence, and transient smearing, especially for high values of the time-scale parameter. In this paper, a new time-scaling algorithm for polyphonic audio signals is described. It uses a multi-scale Gabor analysis for lowfrequency content and a vocoder with phase-locking on transients for the residual signal and for high-frequency content. Compared to a phase-locking vocoder alone, our method significantly reduces both phasiness and transient smearing, especially for high values of the time-scale parameter. For time-contraction (time-scale parameters lower that one), the results seem to be more signaldependant.

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This paper presents a high-quality real-time pitch-shifting algorithm with a time-varying factor for monophonic audio and musical signals. The pitch-shifting algorithm is based on the resampling and time-scale modification method. A new time-scale modification method has been developed which is called the Normalized Filtered Correlation Time-Scale Modification (NFC-TSM) method It uses a ring buffer for time-scaling. The best splicing point is searched in the normalized low-pass filtered signal using the Average Magnitude Difference Function (AMDF). The new method results in low-latency and high-quality pitch-shifting of musical signals.

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We present a method for simulating reverberation in real-time using arbitrary object shapes. This method is an extension of digital plate reverberation where a dry signal is filtered through a physical model of an object vibrating in response to audio input. Using the modal synthesis method, we can simulate the vibration of many different shapes and materials in real time. Sound samples are available at the follwing website: http://cynthia.code404.com/dafx-audio/.

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This article describes an adaptive synthesis technique based on frequency (phase) modulation of arbitrary input signals. The background and motivation for the development of the technique, as well as related work, are discussed. A detailed description of delay line-based phase modulation of sinusoidal and complex signals is provided. The basic design of an implementation of the technique is presented and commented. A series of examples using four different instrumental sources are discussed. The results show a wide range of possible effects through the use of the technique, from addition of higher components, to changes in the odd-even harmonic balance and the introduction of controlled inharmonicity.

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This paper presents a pitch modification scheme, based on the recursive least-squares (RLS) adaptive algorithm, for speech and singing voice signals. The RLS filter is used to determine the linear prediction (LP) model on a sample-by-sample framework, as opposed to the LP-coding (LPC) method, which operates on a block basis. Therefore, an RLS-based approach is able to preserve the natural subtle variations on the vocal tract model, avoiding discontinuities in the synthesized signal and the inherent frame-delay associated to classic methods. The LP residual is modified in the synthesis stage in order to generate the output signal. Listening tests verify the overall quality of the synthesized signal using the RLS approach, indicating that this technique is suitable for realtime applications.

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This article discusses harmonic sinusoid modeling. Unlike standard sinusoid analyzers, the harmonic sinusoid analyzer keeps close watch on partial harmony from an early stage of modeling, therefore guarantees the harmonic relationship among the sinusoids. The key element in harmonic sinusoid modeling is the harmonic sinusoid particle, which can be found by grouping short-time sinusoids. Instead of tracking short-time sinusoids, the harmonic tracker operates on harmonic particles directly. To express harmonic partial frequencies in a compact and robust form, we have developed an inequality-based representation with adjustable tolerance on frequency errors and inharmonicity, which is used in both the grouping and tracking stages. Frequency and amplitude continuity criteria are considered for tracking purpose. Numerical simulations are performed on simple synthesized signals.

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