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
