AES New York 2009
Paper Session P15
P15 - Digital Audio Effects
Sunday, October 11, 2:00 pm — 5:30 pm
Chair: David Berners
P15-1 Discrete Time Emulation of the Leslie Speaker—Jorge Herrera, Craig Hanson, Jonathan S. Abel, Stanford University - Stanford, CA, USA
A discrete-time emulation of the Leslie loudspeaker acoustics is described. The midrange horn and subwoofer baffle are individually modeled, with their rotational dynamics separately tracked, and used to drive time-varying FIR filters applied to the input. The rotational speeds of the horn and baffle are approximated by first-order difference equations having different time constants for acceleration and deceleration. Several time-varying FIR filter methods were explored, all based on impulse responses tabulated over a dense set of horn and baffle angles. In one method, the input sample scales an interpolated impulse response at the current horn or baffle angle, which is added to the output. An example model of a Leslie 44W is presented.
Convention Paper 7925 (Purchase now)
P15-2 A Novel Transient Handling Scheme for Time Stretching Algorithms—Frederik Nagel, Andreas Walther, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Changing either speed or pitch of audio signals without affecting the respective other is often used for music production and creative reproduction, such as remixing. It is also utilized for other purposes such as bandwidth extension and speech enhancement. While stationary signals can be stretched without harming the quality, transients are often not well maintained after time stretching. The present paper demonstrates a novel approach for transient handling in time stretching algorithms. Transient regions are replaced by stationary signals. The thereby removed transients are saved and re-inserted to the time dilated stationary audio signal after time stretching.
Convention Paper 7926 (Purchase now)
P15-3 The Switched Convolution Reverberator—Keun-Sup Lee, Jonathan S. Abel, Stanford University - Stanford, CA, USA; Vesa Välimäki, Helsinki University of Technology - Espoo, Finland; David P. Berners, Universal Audio, Inc. - Santa Cruz, CA, USA
An artificial reverberator having low memory and small computational cost, appropriate for mobile devices, is presented. The reverberator consists of an equalized comb filter driving a convolution with a short noise sequence. The reverberator equalization and decay rate are controlled by low-order IIR filters, and the echo density is that of the noise sequence. While this structure is efficient and readily generates high echo densities, if a fixed noise sequence is used, the reverberator has an unwanted periodicity at the comb filter delay length. To overcome this difficulty, the noise sequence is regularly updated or “switched.” Several structures for updating the noise sequence, including a leaky integrator sensitive to the signal crest factor, and a multi-band architecture, are described.
Convention Paper 7927 (Purchase now)
P15-4 An Emulation of the EMT 140 Plate Reverberator Using a Hybrid Reverberator Structure—Aaron Greenblatt, Stanford University - Stanford, CA, USA; Jonathan S. Abel, David P. Berners, Stanford University - , Stanford, CA, USA, Universal Audio Inc., Scotts Valley, CA, USA
A digital emulation of the Elektromesstechnik (EMT) 140 plate reverberator is presented. The EMT 140 consists of a signal plate and a moveable damping plate; it is approximately linear and time invariant, and its impulse response is characterized by a whip-like onset and high echo density. Here, the hybrid reverberator proposed by Stewart and Murphy, in which a short convolution is run in parallel with a feedback delay network (FDN), is used to model the plate. The impulse response onset is only weakly dependent on the damping control and is modeled by the convolution; the FDN is fit to the impulse response tail. The echo density, equalization, and decay rates are matched at the transition between the convolution and FDN.
Convention Paper 7928 (Purchase now)
P15-5 Simulation of a Guitar Amplifier Stage for Several Triode Models: Examination of Some Relevant Phenomena and Choice of Adapted Numerical Schemes—Ivan Cohen, IRCAM - Paris, France, Orosys R&D, Montpellier, France; Thomas Hélie, IRCAM - Paris, France
This paper deals with the simulation of a high gain triode stage of a guitar amplifier. Triode models taking into account various "secondary phenomena" are considered and their relevance on the stage is analyzed. More precisely, both static and dynamic models (including parasitic capacitances) are compared. For each case, the stage can be modeled by a nonlinear differential algebraic system. For static triode models, standard explicit numerical schemes yield efficient stable simulations of the stage. However, the effect due to the capacitances in dynamic models is audible (Miller effect) and must be considered. The problem becomes stiff and requires the use of implicit schemes. The results are compared for all the models and corresponding VST plug-ins have been implemented.
Convention Paper 7929 (Purchase now)
P15-6 Over-Threshold Power Function Feedback Distortion Synthesis—Tom Rutt, Coast Enterprises, LLC - Asbury Park, NJ, USA
This paper describes an approach to nonlinear distortion synthesis, which uses Over-Threshold Power Function (OTPF) Feedback. The linear gain of an OTPF Feedback distortion synthesizer (using a high gain amplifier) is determined by a linear feedback element. When the output signal becomes greater than a positive threshold value, or less than a negative threshold value, additional OTPF feedback is applied to the distortion synthesizer. The action of this OTPF Feedback Distortion synthesis closely emulates the soft limiting input/output response characteristics of vacuum tube triode grid limit distortion. An important feature of an OTPF feedback distortion synthesizer is that it always behaves as an instantaneous soft limiter and never results in clipping of the output signal peak levels (even at maximum allowable peak input signal levels), if its nonlinear gain constants are set optimally. The paper also describes both circuit and software plug-in realizations of distortion synthesizers that employ Over-Threshold Cubic Function Feedback.
Convention Paper 7930 (Purchase now)
P15-7 Dynamic Panner: An Adaptive Digital Audio Effect for Spatial Audio—Martin Morrell, Joshua D. Reiss, Queen Mary University of London - London, UK
Digital audio effects usually have their parameters controlled by the user, whereas adaptive digital audio effects, or A-DAFx, have some parameters that are driven by the automatic processing and extraction of sound features in the signal content. In this paper we introduce a new A-DAFx, the Dynamic Panner. Based on RMS measurement of its incoming audio signal, the sound source is panned between two user defined points. The audio effect is described and discussed, detailing the technicalities of all the control parameters and the creative context in which the effect can be used. Objective results that can be obtained from the effect are also presented. The audio effect has been implemented as both stereo and 3-dimensional effects using Max/MSP.
Convention Paper 7931 (Purchase now)