AES Vienna 2007: Paper Session P19

P19 - Signal Processing, Sound Quality Design

Monday, May 7, 14:00 — 17:00

Chair: Alfred Kraker

P19-1 Idle Tone Behavior in Sigma Delta Modulation—Enrique Perez Gonzalez, Josh Reiss, Queen Mary, University of London - London, UK
This paper examines the relationship between various unwanted phenomena that plague audio engineers in the design of sigma delta modulators. This work aims to clarify the difference and relationship between DC idle tones, long limit cycles, short limit cycles, and “periodic” short limit cycles, while extending the current knowledge in idle tone behavior. A relationship between the periodic input to the quantizer of a 1-bit delta sigma modulator and the appearance of idle tones is shown. It is shown that for a large range of input signal magnitudes, the fundamental frequency of idle tones is proportional to the DC input. This finding has also been used to examine idle tone aliasing. Numerous simulations are reported which confirm these findings.
Convention Paper 7108 (Purchase now)

P19-2 Low Distortion Sound Reproduction Using 8-Bit uC and ZePoC-Algorithms—Jan Wellmann, Olaf Schnick, Wolfgang Mathis, Leibniz University Hannover - Hannover, Germany
The ZePoC-encoding algorithm for Class-D amplification allows the complete separation of the signal-baseband from all higher-frequency switching artifacts. Real-Time-ZePoC-Encoding demands a lot of computational power, but in applications where recorded signals should be reproduced, they can be encoded by a software-defined ZePoC-System in advance. Reproducing this pre-encoded signal has very low hardware requirements: no digital-analog-converter or linear amplifier is needed; the playback device must only contain memory; a counter for forming the rectangular output-signal; and, if higher output-power is required, an additional switching power-stage and filter. A simple system made up of an 8-bit micro-controller at 16-Mhz clock-rate could reach a signal-to-noise-ratio of 80 dB and a usable frequency range of up to 15 kHz. A test-system made up of an 8-bit RISC-Processor, external memory, and a single-transistor, single power-supply switching-stage will be presented.
Convention Paper 7109 (Purchase now)

P19-3 Efficient, High-Quality Equalization Using a Multirate Filterbank and FIR Filters—Riitta Väänänen, Jarmo Hiipakka, Nokia Research Center - Helsinki, Finland
This paper presents a digital signal processing algorithm for efficient and high-quality audio equalization. In this approach, the original full-band audio signal is first down-sampled and separated into two or more subband signals using a multirate filterbank, after which the equalization is performed in the down-sampled domains. After the equalization, the signal is up-sampled and combined back to a full-band audio signal. Linear phase FIR filters, designed based on the user-controlled parameters, are used to implement the actual equalization. The method presented in this paper helps in designing an implementation that results in computational savings, while still preserving optimal sound quality with any equalization parameter setting.
Convention Paper 7110 (Purchase now)

P19-4 Correction of Crossover Phase Distortion Using Reversed Time All-Pass IIR Filter—Veronique Adam, Sebastien Benz, Goldmund (Audio Networks SA) - Geneva, Switzerland
The purpose of this paper is to describe a correction implementation of group delay distortion arising from a two-way loudspeaker system crossover. Having determined an IIR all-pass filter having a group delay response corresponding to that of the system crossover to be corrected, we have validated under Matlab and implemented in DSP the time reversal solution proposed by S. A. Azizi,. S. R. Powell, and P. M. Chau, enabling an IIR filter to be inversed, while retaining stability and causality. In addition to theory and calculation validation, we have also carried out preliminary listening tests, supporting the evaluation of timber modification, sound clarity, and space localization due to the group delay distortion correction.
Convention Paper 7111 (Purchase now)

P19-5 Natural Timbre in Room Correction Systems—Jan Abildgaard Pedersen, Henrik Green Mortensen, Lyngdorf Audio - Skive, Denmark
Room correction systems are often found to provide a timbre, which is described to be artificial or unnatural. This paper presents a new approach to this problem, which is based on the finding that part of the influence of a listening room is natural to the human ear and should not be removed by a room correction system. More specifically the smooth increase of level toward lower frequencies, also referred to as room gain, must be preserved after applying a room correction system. In the described system this is done as an integral part of the automatic target calculator, which also takes into account the main characteristics of the used loudspeaker, e.g., lower cut-off frequency and directivity index.
Convention Paper 7112 (Purchase now)

P19-6 Multi Core/Multi Thread Processing in Object-Based Real Time Audio Rendering: Approaches and Solutions for an Optimization Problem—Ulrich Reiter, Andreas Partzsch, Technische Universität Ilmenau - Ilmenau, Germany
This paper presents considerations, approaches, and solutions to the problem of optimization of thread distribution for multi core processing in real time audio rendering environments. It explains some basic problems, describes the constraints, and finally suggests an approach based on solving an optimization problem by analyzing a directed graph representing the signal processing flow. The suggested approach can handle an arbitrary number of CPU cores and is therefore well primed for future processor developments.
Convention Paper 7159 (Purchase now)