AES San Francisco 2012
Poster Session P3
P3 - Audio Effects and Physical Modeling
Friday, October 26, 10:00 am — 11:30 am (Foyer)
P3-1 Luciverb: Iterated Convolution for the Impatient—Jonathan S. Abel, Stanford University - Stanford, CA, USA; Michael J. Wilson, Stanford University - Stanford, CA, USA
An analysis of iteratively applied room acoustics used by Alvin Lucier to create his piece "I'm Sitting in a Room" is presented, and a real-time system allowing interactive control over the number of rooms in the processing chain is described. Lucier anticipated that repeated application of a room response would bring out room resonances and smear the input sound over time. What was unexpected was the character of the smearing, turning a transient input into a sequence of crescendos at the room modes, ordered from high-frequency to low-frequency. Here, a room impulse response convolve with itself L times is shown have energy at the room mofes, each with a roughly Gaussian envelope, peaking at the observed L/2 times the frequency-dependent decay time.
Convention Paper 8691 (Purchase now)
P3-2 A Tilt Filter in a Servo Loop—John Lazzaro, University of California, Berkeley - Berkeley, CA, USA; John Wawrzynek, University of California, Berkeley - Berkeley, CA, USA
Tone controls based on the tilt filter first appeared in 1982, in the Quad 34 Hi-Fi preamp. More recently, tilt filters have found a home in specialist audio processors such as the Elysia mpressor. This paper describes a novel dynamic filter design based on a tilt filter. A control system sets the tilt slope of the filter, in order to servo the spectral median of the filter output to a user-specified target. Users also specify a tracking time. Potential applications include single-instrument processing (in the spirit of envelope filters) and mastering (for subtle control of tonal balance). Although we have prototyped the design as an AudioUnit plug-in, the architecture is also a good match for analog circuit implementation.
Convention Paper 8692 (Purchase now)
P3-3 Multitrack Mixing Using a Model of Loudness and Partial Loudness—Dominic Ward, Birmingham City University - Birmingham, UK; Joshua D. Reiss, Queen Mary University of London - London, UK; Cham Athwal, Birmingham City University - Birmingham, UK
A method for generating a mix of multitrack recordings using an auditory model has been developed. The proposed method is based on the concept that a balanced mix is one in which the loudness of all instruments are equal. A sophisticated psychoacoustic loudness model is used to measure the loudness of each track both in quiet and when mixed with any combination of the remaining tracks. Such measures are used to control the track gains in a time-varying manner. Finally we demonstrate how model predictions of partial loudness can be used to counteract energetic masking for any track, allowing the user to achieve better channel intelligibility in complex music mixtures.
Convention Paper 8693 (Purchase now)
P3-4 Predicting the Fluctuation Strength of the Output of a Spatial Chorus Effects Processor—William L. Martens, University of Sydney - Sydney, NSW, Australia; Robert W. Taylor, University of Sydney - Sydney, NSW, Australia; Luis Miranda, University of Sydney - Sydney, NSW, Australia
The experimental study reported in this paper was motivated by an exploration of a set of related audio effects comprising what has been called “spatial chorus.” In contrast to a single-output, delay-modulation-based effects processor that produces a limited range of results, complex spatial imagery is produced when parallel processing channels are subjected to incoherent delay modulation. In order to develop a more adequate user interface for control of such “spatial chorus” effects processing, a systematic investigation of the relationship between algorithmic parameters and perceptual attributes was undertaken. The starting point for this investigation was to perceptually scale the amount of modulation present in a set of characteristic stimuli in terms of the auditory attribute that Fastl and Zwicker called “fluctuation strength.”
Convention Paper 8694 (Purchase now)
P3-5 Computer-Aided Estimation of the Athenian Agora Aulos Scales Based on Physical Modeling—Areti Andreopoulou, New York University - New York, NY, USA; Agnieszka Roginska, New York University - New York, NY, USA
This paper presents an approach to scale estimation for the ancient Greek Aulos with the use of physical modeling. The system is based on manipulation of a parameter set that is known to affect the sound of woodwind instruments, such as the reed type, the active length of the pipe, its inner and outer diameters, and the placement and size of the tone-holes. The method is applied on a single Aulos pipe reconstructed from the Athenian Agora fragments. A discussion follows on the resulting scales and the system’s advantages, and limitations.
Convention Paper 8695 (Purchase now)
P3-6 A Computational Acoustic Model of the Coupled Interior Architecture of Ancient Chavín—Regina E. Collecchia, Stanford University - Stanford, CA, USA; Miriam A. Kolar, Stanford University - Stanford, CA, USA; Jonathan S. Abel, Stanford University - Stanford, CA, USA
We present a physical, modular computational acoustic model of the well-preserved interior architecture at the 3,000-year-old Andean ceremonial center Chavín de Huántar. Our previous model prototype [Kolar et. al. 2010] translated the acoustically coupled topology of Chavín gallery forms to a model based on digital waveguides (bi-directional by definition), representing passageways, connected through reverberant scattering junctions, representing the larger room-like areas. Our new approach treats all architectural units as “reverberant” digital waveguides, with scattering junctions at the discrete planes defining the unit boundaries. In this extensible and efficient lumped-element model, we combine architectural dimensional and material data with sparsely measured impulse responses to simulate multiple and circulating arrival paths between sound sources and listeners.
Convention Paper 8696 (Purchase now)
P3-7 Simulating an Asymmetrically Saturated Nonlinearity Using an LNLNL Cascade—Keun Sup Lee, DTS, Inc. - Los Gatos, CA, USA; Jonathan S. Abel, Stanford University - Stanford, CA, USA
The modeling of a weakly nonlinear system having an asymmetric saturating nonlinearity is considered, and a computationally efficient model is proposed. The nonlinear model is the cascade of linear filters and memoryless nonlinearities, an LNLNL system. The two nonlinearities are upward and downward saturators, limiting, respectively, the amplitude of their input for either positive or negative excursions. In this way, distortion noted in each half an input sinusoid can be separately controlled. This simple model is applied toy simulating the signal chain of the Echoplex EP-4 tape delay, where informal listening tests showed excellent agreement between recorded and simulated program material.
Convention Paper 8697 (Purchase now)
P3-8 Coefficient Interpolation for the Max Mathews Phasor Filter—Dana Massie, Audience, Inc. - Mountain View, CA, USA
Max Mathews described what he named the “phasor filter,” which is a flexible building block for computer music, with many desirable properties. It can be used as an oscillator or a filter, or a hybrid of both. There exist analysis methods to derive synthesis parameters for filter banks based on the phasor filter, for percussive sounds. The phasor filter can be viewed as a complex multiply, or as a rotation and scaling of a 2-element vector, or as a real valued MIMO (multiple-input, multiple-output) 2nd order filter with excellent numeric properties (low noise gain). In addition, it has been proven that the phasor filter is unconditionally stable under time varying parameter modifications, which is not true of many common filter topologies. A disadvantage of the phasor filter is the cost of calculating the coefficients, which requires a sine and cosine in the general case. If pre-calculated coefficients are interpolated using linear interpolation, then the poles follow a trajectory that causes the filter to lose resonance. A method is described to interpolate coefficients using a complex multiplication that preserves the filter resonance.
Convention Paper 8698 (Purchase now)
P3-9 The Dynamic Redistribution of Spectral Energies for Upmixing and Re-Animation of Recorded Audio—Christopher J. Keyes, Hong Kong Baptist University - Kowloon, Hong Kong
This paper details a novel approach to upmixing any n channels of audio to any arbitrary n+ channels of audio using frequency-domain processing to dynamically redistribute spectral energies across however many channels of audio are available. Although primarily an upmixing technique, the process may also help the recorded audio regain the sense of “liveliness” that one encounters in concerts of acoustic music, partially mimicking the effects of sound spectra being redistributed throughout a hall due to the dynamically changing radiation patterns of the instruments and the movements of the instruments themselves, during performance and recording. Preliminary listening tests reveal listeners prefer this technique 3 to 1 over a more standard upmixing technique.
Convention Paper 8699 (Purchase now)
P3-10 Matching Artificial Reverb Settings to Unknown Room Recordings: A Recommendation System for Reverb Plugins—Nils Peters, International Computer Science Institute - Berkeley, CA, USA; University of California Berkeley - Berkeley, CA, USA; Jaeyoung Choi, International Computer Science Institute - Berkeley, CA, USA; Howard Lei, International Computer Science Institute - Berkeley, CA, USA
For creating artificial room impressions, numerous reverb plugins exist and are often controllable by many parameters. To efficiently create a desired room impression, the sound engineer must be familiar with all the available reverb setting possibilities. Although plugins are usually equipped with many factory presets for exploring available reverb options, it is a time-consuming learning process to find the ideal reverb settings to create the desired room impression, especially if various reverberation plugins are available. For creating a desired room impression based on a reference audio sample, we present a method to automatically determine the best matching reverb preset across different reverb plugins. Our method uses a supervised machine-learning approach and can dramatically reduce the time spent on the reverb selection process.
Convention Paper 8700 (Purchase now)
