Paper Archive

We propose a new style of continuous-time filter design composed of a cascade of 2nd-order state variable filters (SVFs) and a global feedback path. This family of filters is parameterized by the SVF cutoff frequencies and resonances, as well as the global feedback amount. For the case of two identical SVFs in cascade and a specific value of the SVF resonance, the proposed design reduces to the well-known Moog ladder filter. For another resonance value, it approximates the Octave CAT filter. The resonance parameter can be used to create new filters as well. We study the pole loci and transfer functions of the SVF building block and entire filter. We focus in particular on the effect of the proposed parameterization on important aspects of the filter’s response, including the passband gain and cutoff frequency error. We also present the first in-depth study of the Octave CAT filter circuit.

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Artificial reverberation algorithms generally imitate the frequency-dependent decay of sound in a room quite inaccurately. Previous research suggests that a 5% error in the reverberation time (T60) can be audible. In this work, we propose to use an accurate graphic equalizer as the attenuation filter in a Feedback Delay Network reverberator. We use a modified octave graphic equalizer with a cascade structure and insert a high-shelf filter to control the gain at the high end of the audio range. One such equalizer is placed at the end of each delay line of the Feedback Delay Network. The gains of the equalizer are optimized using a new weighting function that acknowledges nonlinear error propagation from filter magnitude response to reverberation time values. Our experiments show that in real-world cases, the target T60 curve can be reproduced in a perceptually accurate manner at standard octave center frequencies. However, for an extreme test case in which the T60 varies dramatically between neighboring octave bands, the error still exceeds the limit of the just noticeable difference but is smaller than that obtained with previous methods. This work leads to more realistic artificial reverberation.

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This papers deals with a 3-D audio system that has been developed for desktop multimedia environments. The system has the ability to place virtual sources at arbitrary azimuths and elevations around the listener’s head based on HRTF binaural synthesis. A listener seated in front of a computer and two loudspeakers placed at each side of the monitor have been considered. Transaural reproduction using loudspeakers has been used for rendering the sound field to listener ears. Furthermore the system can cope with slight movements of the listener head. Head position is monitored by means of a simple computer vision algorithm. Four head position coordinates (x,y,z,φ) in order to allow free movements of the listener are continuously estimated. Cross-talk cancellation filters and virtual sources locations are updated depending on these head coordinates.

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Iterative phase formulations allow for the generalization of many oscillatory sound synthesis methods from circles to general parametrizable loops, with or without explicit geometric contexts. This paper describes this approach, leading to the ability to perform modulation, feedback and chaotic oscillations over deformed circles that can include ill-behaved geometries, while allowing modulations or feedback to be deformed as well.

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Memoryless waveshapers are commonly used in audio signal processing. In discrete time, they suffer from well-known aliasing artifacts. We present a method for applying antiderivative antialising (ADAA), which mitigates aliasing, to any waveshaping function that can be represented as a piecewise polynomial. Specifically, we treat the special case of a piecewise linear waveshaper. Furthermore, we introduce a method for for replacing the sharp corners and jump discontinuities in any piecewise linear waveshaper with smoothed polynomial approximations, whose derivatives match the adjacent line segments up to a specified order. This piecewise polynomial can again be antialiased as a special case of the general piecewise polynomial. Especially when combined with light oversampling, these techniques are effective at reducing aliasing and the proposed method for rounding corners in piecewise linear waveshapers can also create more “realistic” analog-style waveshapers than standard piecewise linear functions.

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A previously existing nonlinear differential equation system modeling the EMS VCS3 voltage controlled filter is reformulated here in polynomial form, avoiding the expensive computation of transcendent functions imposed by the original model. The new system is discretized by means of an implicit numerical scheme, and solved using Newton-Raphson iterations. While maintaining instantaneous controllability, the algorithm is both significantly faster and more accurate than the previous filter-based solution. A real time version of the model has been implemented under the PureData audio processing environment and as a VST plugin.

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In this paper we consider the problem of separating different sound sources in multichannel audio signals. Different approaches to the problem of Blind Source Separation (BSS), e.g. the Independent Component Analysis (ICA) originally proposed by Herault and Jutten, and extensions to this including delays, work fine for artificially mixed signals. However the quality of the separated signals is severely degraded for real sound recordings when there is reverberation. We consider the system with 2 sources and 2 sensors, and show how we can improve the quality of the separation by a simple model of the audio scene. More specifically we estimate the delays between the sensor signals, and put constraints on the deconvolution coefficients.

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We present a novel mapping technique for sound synthesis. The technique extends the familiar concept of morphing to the domain of synthesis parameters. A morph between defined points in the parameter space representing desirable sounds is itself parameterized with high-level controls. The choice of end points of the morph and the extent of the morph are used as input handles to map arbitrary control signals to the synthesis parameters. Additional off-line methods control the interpolation functions and selection of parameter points. We discuss a tool to allow creation, manipulation and usage of such mappings.

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A central challenge for a cappella singers is to adjust their intonation and to stay in tune relative to their fellow singers. During editing of a cappella recordings, one may want to adjust local intonation of individual singers or account for global intonation drifts over time. This requires applying a time-varying pitch-shift to the audio recording, which we refer to as adaptive pitch-shifting. In this context, existing (semi-)automatic approaches are either laborintensive or face technical and musical limitations. In this work, we present automatic methods and tools for adaptive pitch-shifting with applications to intonation adjustment in a cappella recordings. To this end, we show how to incorporate time-varying information into existing pitch-shifting algorithms that are based on resampling and time-scale modification (TSM). Furthermore, we release an open-source Python toolbox, which includes a variety of TSM algorithms and an implementation of our method. Finally, we show the potential of our tools by two case studies on global and local intonation adjustment in a cappella recordings using a publicly available multitrack dataset of amateur choral singing.

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Antiderivative Antialiasing (ADAA), has become a pivotal method for reducing aliasing when dealing with nonlinear function at audio rate. However, its implementation requires analytical computation of the antiderivative of the nonlinear function, which in practical cases can be challenging without a symbolic solver. Moreover, when the nonlinear function is given by measurements it must be approximated to get a symbolic description. In this paper, we propose a simple approach to ADAA for practical applications that employs numerical integration of lookup tables (LUTs) to approximate the antiderivative. This method eliminates the need for closed-form solutions, streamlining the ADAA implementation process in industrial applications. We analyze the trade-offs of this approach, highlighting its computational efficiency and ease of implementation while discussing the potential impact of numerical integration errors on aliasing performance. Experiments are conducted with static nonlinearities (tanh, a simple wavefolder and the Buchla 259 wavefolding circuit) and a stateful nonlinear system (the diode clipper).

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