Abstract

The paper presents two new approaches to artefact-free real-time updates of the impulse response in convolution. Both approaches are based on incremental updates of the filter. This can be useful for several applications within digital audio processing: parametrisation of convolution reverbs, dynamic filters, and live convolution. The development of these techniques has been done within the framework of a research project on crossadaptive audio processing methods for live performance. Our main motivation has thus been live convolution, where the signals from two music performers are convolved with each other, allowing the musicians to “play through each other’s sound”.

Highlights

  • Convolution has been used for filtering, reverberation, spatialisation, and as a creative tool for cross-synthesis in a variety of contexts [1,2,3,4,5]

  • Time-varying convolution has been explored in continuous-time systems [9], and in discrete-time systems, both finite impulse response (FIR) and infinite impulse response (IIR) coefficient-modulated filters have been extensively discussed in [10]

  • Since this algorithm depends on the circular property of the discrete Fourier transform (DFT), which cannot be guaranteed with a fully time-varying impulse response, it cannot be used in a practical implementation of the TVFIR described by Equations (24)–(26), This is because the Overlap-save algorithm (OLS) algorithm expects that the impulse response data will not vary over the duration of the convolution, which is not the case if both signals are continuously varying

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Summary

Introduction

Convolution has been used for filtering, reverberation, spatialisation, and as a creative tool for cross-synthesis in a variety of contexts [1,2,3,4,5]. One of the methods presented here have been discussed by two of the authors in an earlier conference paper [8], where artefact-free updates of the filter allowed live sampling of the impulse response. The current paper expands on this work by adding another method for time-varying filter coefficients Both methods for filter updates are based on a similar concept, where coefficients are replaced at audio rate, enabling artefact free transition from old to new filter coefficients, even with arbitrary different coefficients. A number of practical experiments in studio and live sessions have been done during the time since the conference paper was written Reflections on these artistic explorations have been included in the present article

Time-Varying Filters
Time-Varying Finite Impulse Response Filters
Dynamic Replacement of Impulse Responses
Time-Varying Convolution
Fixing Coefficients
Test Signals
Implementation
Direct Convolution
Fast Convolution
Partitioned Convolution
Csound Opcodes
Liveconvolver
TV Convolver
Demo Sounds
Future Work
Conclusions
Full Text
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