AES Munich 2009
Paper Session P27
Sunday, May 10, 13:00 — 17:00
Chair: Günther Thiele
P27-1 On the Myth of Pulse Width Modulated Spectrum in Theory and Practice—Arnold Knott, Technical University of Denmark - Lyngby, Denmark, Harman/Becker Automotive Systems GmbH, Straubing, Germany; Gerhard Pfaffinger, Harman/Becker Automotive Systems GmbH - Straubing, Germany; Michael A. E. Andersen, Technical University of Denmark - Lyngby, Denmark
Switch-mode audio power amplifiers are commonly used in sound reproduction. Their well-known drawback is the radiation of high frequent energy, which can disturb radio and TV receivers. The designer of switch-mode audio equipment therefore need to make arrangements to prevent this coupling, which would otherwise result in bad audio performance. A deep understanding of the pulse width modulated (PWM) signal is therefore essential, which resulted in different mythic models as pulse, trapezoidal, or Double Fourier Series (DFS) representations in the past. This paper will clarify these theoretical approaches by comparing them with reality from both the time and the frequency domain perspective. For validation a switch-mode audio power amplifier was built, delivering the contents material with less than 0.06 percent distortion across the audio band at 50 W. The switch-mode signals have been evaluated very precisely in time and spectral domain to enlighten the assumptions about the PWM spectra and decrypt this myth.
Convention Paper 7799 (Purchase now)
P27-2 Design Approaches for Psychoacoustical In-Band Noise Shaping Filters—Jochen Hahn, University of Kaiserslautern - Kaiserslautern, Germany
Noise shaping is a state-of-the-art technique to preserve the perceived quality of audio signals when requantization happens. Noise shaping filters are special filters because of the nonlinear characteristics in hearing. They have to be taken into account when designing these special digital audio filters. The design approaches presented in this contribution meet these requirements. They minimize or limit the filter magnitude, the unweighted noise amplification, and the group delay characteristics of the filter.
Convention Paper 7800 (Purchase now)
P27-3 A New Analog Input Topology for Extreme Dynamic Range Analog to Digital Conversion—Jamie Angus, University of Salford - Salford, Greater Manchester, UK
The purpose of this paper is to introduce a novel form of input topology for the analog inputs of oversampled analog to digital converters. This new topology, when used with existing components, can achieve a dynamic range of 28 linear bits but has the potential to achieve even more if suitable technology can be developed. The paper analyzes the current limitations of existing topologies, presents the new topology, and shows how it can achieve much higher dynamic ranges. The optimal application of the topology and means of extending it for higher dynamic ranges is also discussed.
Convention Paper 7801 (Purchase now)
P27-4 Automatic Equalization of Flat TV Loudspeakers Using Parametric IIR Filters—Herwig Behrends, NXP Semiconductors - Hamburg, Germany; Adrian von dem Knesebeck, Helmut Schimidt University, University of the Federal Armed Forces - Hamburg, Germany; Werner Bradinal, Peter Neumann, NXP Semiconductors - Hamburg, Germany; Udo Zölzer, Helmut Schmidt University, University of the Federal Armed Forces - Hamburg, Germany
Small loudspeakers used in the today’s flat television set cabinets and the requirement for “invisible sound” lead to a frequency response that is influenced in a very disadvantageous way by the physical design constraints. Loudspeakers are deeply embedded within the cabinet—the sound is thus forced through narrow vents, funnels or other waveguides. Down- or backfiring placements of the loudspeakers are also common practice, in order to minimize their visibility as much as possible. Generally, this leads to a non-flat frequency response with a strong coloration of the sound. We present an approach to compensate these effects by means of simple second order equalizer sections (biquads), where center frequency, gain, and bandwidth of the equalizer sections are automatically calculated from a measured frequency response. The tool is usable in a laboratory environment, with relatively inexpensive standard PC sound cards and microphones.
Convention Paper 7802 (Purchase now)
P27-5 Audio n-Genie: Domain Specific Language for Audio Processing—Tiziano Leidi, Institute for Applied Computer Science and Industrial Technologies of Southern Switzerland (ICIMSI) - Manno, Switzerland; Thierry Heeb, ANAGRAM Technologies SA - Préverenges, Switzerland; Marco Colla, Institute for Applied Computer Science and Industrial Technologies of Southern Switzerland (ICIMSI) - Manno, Switzerland; Jean-Philippe Thiran, Ecole Polytechnique Federale de Lausanne (EPFL) - Lausanne, Switzerland
Specialized development environments represent today an important added value for domain specific system providers suffering the lack of a dedicated, ergonomic, efficient, and affordable tool able to boost their core business. This paper describes Audio n-genie, a domain-specific language and its associated development environment supporting the automatic production, by mean of component-based model-driven generative programming, of digital audio processing applications.
Convention Paper 7803 (Purchase now)
P27-6 Acoustic Echo Cancellation Using MIMO Blind Deconvolution—Ephraim Gower, Malcolm Hawksford, University of Essex - Colchester, UK
A new multiple-input multiple-output frame-processing algorithm is introduced that exploits blind deconvolution for acoustic echo cancellation. The channel deconvolution filters, which can be blindly estimated as either finite impulse or infinite impulse responses, are optimized by maximizing the information flow through several nonlinear neurons. The algorithm requires that for every system audio output there be a corresponding microphone for effective feedback signal separation/cancellation. The desired talker signal from the algorithm outputs is recognized and transmitted while the identified feedback signals are discarded.
Convention Paper 7804 (Purchase now)
P27-7 Implementing Audio Algorithms and Integrating Processor-Specific Code Using Model Based Design—Arvind Ananthan, The MathWorks - Natick, MA, USA; Mark Corless, The MathWorks - Novi, MI, USA; Marco Roggero, The MathWorks - Ismaning, Germany
This paper explores the final stages in the model-based design and implementation of an audio algorithm on a fixed-point embedded processor. Once the algorithm, a 3-band parametric equalizer in this example, is designed and simulated using a combination of scripting and graphical modeling tools, real-time C-code is automatically generated from this model. This paper illustrates how algorithmic C-code generated from such a model in Simulink can be integrated into a stand-alone embedded project as a library and implemented on an Analog Devices Blackfin® 537 processor. It also elaborates how processor specific C-callable assembly code can then be integrated into the model for both simulation and code generation to improve its execution performance on this processor.
Convention Paper 7805 (Purchase now)
P27-8 Subjective and Objective Evaluation of the Audio Vacuum-Tube Amplifiers—Andrzej Dobrucki, Wroclaw University of Technology - Wroclaw, Poland; Stanislaw Maleczek, Military Institute of Engineering Technology - Wroclaw, Poland; Maurycy Kin, Wroclaw University of Technology - Wroclaw, Poland
The subjective and objective evaluation of 5 high-quality vacuum-tube audio amplifiers is presented in this paper. As the reference the professional transistor amplifier has been used. The subjective evaluation has been done by the team of judges as well as with the computer-based psychoacoustic model according with PAQM protocol. The amplifiers’ sound quality assessed by the listeners is consistent with the one evaluated with the use of the psychoacoustic model. It was found that the best sound quality is obtained by vacuum-tube amplifiers, the worst—by the reference amplifier. The results of subjective evaluation are inconsistent with quality assessed by measurement of objective parameters: all amplifiers have comparable quality, but the best is the transistor amplifier because of lowest level of THD+N level.
Convention Paper 7806 (Purchase now)