AES New York 2015
Paper Session P1
P1 - Signal Processing
Thursday, October 29, 9:00 am — 12:30 pm (Room 1A08)
Chair:
Scott Norcross, Dolby Laboratories - San Francisco, CA, USA
P1-1 Time-Frequency Analysis of Loudspeaker Sound Power Impulse Response—Pascal Brunet, Samsung Research America - Valencia, CA USA; Audio Group - Digital Media Solutions; Allan Devantier, Samsung Research America - Valencia, CA, USA; Adrian Celestinos, Samsung Research America - Valencia, CA, USA
In normal conditions (e.g., a living room) the total sound power emitted by the loudspeaker plays an important role in the listening experience. Along with the direct sound and first reflections, the sound power defines the loudspeaker performance in the room. The acoustic resonances of the loudspeaker system are especially important, and thanks to spatial averaging, are more easily revealed in the sound power response. In this paper we use time-frequency analysis to study the spatially averaged impulse response and reveal the structure of its resonances. We also show that the net effect of loudspeaker equalization is not only the attenuation of the resonances but also the shortening of their duration.
Convention Paper 9354 (Purchase now)
P1-2 Low-Delay Transform Coding Using the MPEG-H 3D Audio Codec—Christian R. Helmrich, International Audio Laboratories - Erlangen, Germany; Michael Fischer, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Recently the ISO/IEC MPEG-H 3D Audio standard for perceptual coding of one or more audio channels has been finalized. It is a little-known fact that, particularly for communication applications, the 3D Audio core-codec can be operated in a low-latency configuration in order to reduce the algorithmic coding/decoding delay to 44, 33, 24, or 18 ms at a sampling rate of 48 kHz. This paper introduces the essential coding tools required for high-quality low-delay coding–transform splitting, intelligent gap filling, and stereo filling–and demonstrates by means of blind listening tests that the achievable subjective performance compares favorably with, e.g., that of HE-AAC even at low bit-rates.
Convention Paper 9355 (Purchase now)
P1-3 Dialog Control and Enhancement in Object-Based Audio Systems—Jean-Marc Jot, DTS, Inc. - Los Gatos, CA, USA; Brandon Smith, DTS, Inc. - Bellevue, WA, USA; Jeff Thompson, DTS, Inc. - Bellevue, WA, USA
Dialog is often considered the most important audio element in a movie or television program. The potential for artifact-free dialog salience personalization is one of the advantages of new object-based multichannel digital audio formats, along with the ability to ensure that dialog remains comfortably audible in the presence of concurrent sound effects or music. In this paper we review some of the challenges and requirements of dialog control and enhancement methods in consumer audio systems, and their implications in the specification of object-based digital audio formats. We propose a solution incorporating audio object loudness metadata, including a simple and intuitive consumer personalization interface and a practical head-end encoder extension.
Convention Paper 9356 (Purchase now)
P1-4 Frequency-Domain Parametric Coding of Wideband Speech–A First Validation Model—Aníbal Ferreira, University of Porto - Porto, Portugal; Deepen Sinha, ATC Labs - Newark, NJ, USA
Narrow band parametric speech coding and wideband audio coding represent opposite coding paradigms involving audible information, namely in terms of the specificity of the audio material, target bit rates, audio quality, and application scenarios. In this paper we explore a new avenue addressing parametric coding of wideband speech using the potential and accuracy provided by frequency-domain signal analysis and modeling techniques that typically belong to the realm of high-quality audio coding. A first analysis-synthesis validation framework is described that illustrates the decomposition, parametric representation, and synthesis of perceptually and linguistically relevant speech components while preserving naturalness and speaker specific information.
Convention Paper 9357 (Purchase now)
P1-5 Proportional Parametric Equalizers—Application to Digital Reverberation and Environmental Audio Processing—Jean-Marc Jot, DTS, Inc. - Los Gatos, CA, USA
Single-band shelving or presence boost/cut filters are useful building blocks for a wide range of audio signal processing functions. Digital filter coefficient formulas for elementary first- or second-order IIR parametric equalizers are reviewed and discussed. A simple modification of the classic Regalia-Mitra design yields efficient solutions for tunable digital equalizers whose dB magnitude frequency response is proportional to the value of their gain control parameter. Practical applications to the design of tone correctors, artificial reverberators and environmental audio signal processors are described.
Convention Paper 9358 (Purchase now)
P1-6 Comparison of Parallel Computing Approaches of a Finite-Difference Implementation of the Acoustic Diffusion Equation Model—Juan M. Navarro, UCAM - Universidad Católica San Antonio - Guadalupe (Murcia), Spain; Baldomero Imbernón, UCAM Catholic University of San Antonio - Murcia, Spain; José J. López, Universitat Politcnica de Valencia - Valencia, Spain; José M. Cecilia, UCAM Catholic University of San Antonio - Murcia, Spain
The diffusion equation model has been intensively researched as a room-acoustics simulation algorithm during last years. A 3-D finite-difference implementation of this model was proposed to evaluate the propagation over time of sound field within rooms. Despite the computational saving of this model to calculate the room energy impulse response, elapsed times are still long when high spatial resolutions and/or simulations in several frequency bands are needed. In this work several data-parallel approaches of this finite-difference solution on Graphics Processing Units are proposed using a compute unified device architecture programming model. A comparison of their performance running on different models of Nvidia GPUs is carried out. In general, 2D vertical block approach running in a Tesla K20C shows the best speed-up of more than 15 times versus CPU version.
Convention Paper 9359 (Purchase now)
P1-7 An Improved and Generalized Diode Clipper Model for Wave Digital Filters—Kurt James Werner, Center for Computer Research in Music and Acoustics (CCRMA) - Stanford, CA, USA; Stanford University; Vaibhav Nangia, Stanford University - Stanford, CA, USA; Alberto Bernardini, Politecnico di Milano - Milan, Italy; Julius O. Smith, III, Stanford University - Stanford, CA, USA; Augusto Sarti, Politecnico di Milano - Milan, Italy
We derive a novel explicit wave-domain model for “diode clipper" circuits with an arbitrary number of diodes in each orientation, applicable, e.g., to wave digital filter emulation of guitar distortion pedals. Improving upon and generalizing the model of Paiva et al. (2012), which approximates reverse-biased diodes as open circuits, we derive a model with an approximated correction term using two Lambert W functions. We study the energetic properties of each model and clarify aspects of the original derivation. We demonstrate the model's validity by comparing a modded Tube Screamer clipping stage emulation to SPICE simulation.
Convention Paper 9360 (Purchase now)
