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Session K Monday, May 14 13:30 - 17:30 hr Room B

Signal Processing for Audio, Part 1

Chair: Peter Eastty, Sony Pro-Audio R&D, Oxford, UK

13:30 hr K-1
Analysis, Design and Assessment of Class A, B, AB, G and H Audio Power Amplifier Output Stages Based on Matlab Software
Rui Seara, Sidnei Noceti Filho & Rosalfonso Bortoni
Federal University of Santa Catarina, Florian█polis, Brazil

A procedure for analyzing, designing and assessing audio power amplifier output stages operating in class A, B, AB, G and H with reactive loads is presented. This study considers steady-state sinusoidal analysis for BJT, IGBT and MOSFET technologies. Electrical-mechanical-acoustical models of loudspeakers and enclosures are used whose parameters are obtained through the Thiele-Small model. An equivalent electrical-thermal model for the transistor-heatsink-ambience associated with the instantaneous and average powers is used for designing the power stage. A MATLAB software has been developed, which provides a considerable support to the designer for all required phases in an audio power amplifier output stage design.
Paper 5358

14:00 hr K-2
Not-Linear Convolution: A New Approach for the Auralization of Distorting Systems
Angelo Farina, Alberto Bellini & Enrico Armelloni
University of Parma, Parma, Italy

This work defines a new method for processing audio signals, with the aim to recreate an audible simulation (auralization) of the modification imposed on the original signal by a complex system. The new method is the extension of the classic auralization process based on the linear convolution of the "dry" original signal with the impulse response of the system. The extension allows for the emulation of not-linear systems, characterized in terms of harmonic distortion at several orders. The work first presents the mathematical framework of the proposed implementation, then it is shown how a not- linear system can be experimentally characterized by a new measurement method of multiple impulse responses at various harmonic orders. Finally it is shown how these impulse responses can be employed in a multiple convolution process: an experimental demonstration is given of the similarity of the numerically processed sound with the live recording coming from a highly distorting device.
Paper 5359

14:30 hr K-3
Pre- and De-Emphasis - A Forgotten Necessity
George Brock-Nannestad
Patent Tactics, Gentofte, Denmark

A number of physical limitations are inherent to analog recording media, and for this reason compromises have to be accepted. In order to widen the frequency range of recordings, mechanical, optical, and magnetic recording have used pre-emphasis at recording which is then suppressed again by a complementary de-emphasis at replay. The paper traces the parallel development in all three fields of analog recording.
Paper 5360

15:00 hr K-4
A Distortion-Free PWM Coder for All-Digital Audio Amplifiers
Andreas Floros, Nicolaos Tatlas & John Mourjopoulos
University of Patras, Patras, Greece

Using analytic PCM-to-PWM mapping, combined with a novel method for eliminating PWM-induced distortions (Jithering), a distortion-free, all-digital and high-quality PWM coder was developed. High efficiency and performance is achieved at switching frequencies between 44.1-176.4kHz. A Field Programmable Gate Array-based environment was used for the implementation of the PWM converter, which is suitable for any digital audio applications.
Paper 5361

15:30 hr K-5
Smart Directional and Diffuse Digital Loudspeaker Arrays
Malcolm Hawksford
University of Essex, Colchester, UK

A theory of smart loudspeaker arrays is described where a modified Fourier technique yields complex filter coefficients to determine the broadband radiation characteristics of a uniform array of micro drive units. Beam width and direction are individually programmable over a 180-degree arc, where multiple agile and steerable beams carrying dissimilar signals can be accommodated. A novel method of diffuse filter design is also presented that endows the directional array with diffuse radiation properties.
Paper 5362

16:00 hr K-6
Optimized DSP Implementation of Non-Linear Quantization
Raghunath Rao & Girish Subramaniam
Cirrus Logic Inc, Austin, TX, USA

Non-linear quantization of the type INT(x^(M/N) + constant) is commonly used in audio compression techniques, particularly MPEG-1 and MPEG-2 layer III (MP3) and MPEG Advanced Audio Coding (AAC). Finding a suitable DSP implementation is a problem since lookup table methods are prohibitive due to excessive storage requirements, conventional series approximation methods do not give sufficient precision, and not all processors have log/exp assist functions. This paper describes a method which utilizes the property of geometric periodicity of the x^(M/N) function to first normalize the problem to a small range of input x. Subsequently one can choose to perform the x^(M/N) in this limited range based on lookup, interpolation, or series expansion, and finally re-normalize the output to obtain the overall answer. Using a hybrid scheme based on lookup and interpolation very good overall precision is achieved. Compared to direct application of any of the above techniques, there is very little additional computational burden, and the improvement in precision is very significant. Mathematically, this method is shown to be a special case of log-exp based computation where the log is quantized.
Paper 5363

16:30 hr K-7
Sound Equalization in a Noisy Environment
Meir Tzur (Zibulski) & Alexander Goldin
BIT Innovation Technologies, Tirat HaCarmel, Israel

This paper addresses the problem of equalizing an audio signal in a constantly changing noisy environment. The purpose of equalization is to provide perceptually equal loudness of sound regardless of the environmental conditions. Based on an automatic estimation of noise level and its spectral content, selective amplification of frequencies masked by noise is performed. In the case of speech signals, the result is intelligible speech regardless of the surrounding noise. For musical signals, an improved comprehension of the musical content is achieved.
Paper 5364

17:00 hr K-8
Zero Position Coding (ZePoC) - A Generalized Concept of Pulse-Length Modulated Signals and its Application to Class-D Audio Power Amplifiers
Martin Streitenberger (1), Frank Felgenhauer (1), Helmut Bresch (1) & Wolfgang Mathis (2)
University of Magdeburg, Magdeburg, Germany
University of Hannover, Hannover, Germany

Zero Position Coding (ZePoC) is introduced in this paper as a generalized concept for describing methods of generating binary signals with varying pulse-lengths. This class of signals is of basic interest within concepts of class-D power amplification. It is emphasized that from a generalized point of view such signals are generated by coding the positions of the zero-crossings (sign-changes) of some auxiliary signal being uniquely determined by the audio input signal. The new ZePoC concept is shown to include classical methods like NPWM and UPWM as well as a new method, SB-ZePoC, which allows the generation of a binary signal with separated base band. The methods are compared showing that SB-ZePoC is favored for use in the class-D amplification concept. Results of a first full audio band implementation of SB-ZePoC are given.
Paper 5365

 

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