Last Updated: 20050816, mei
Saturday, October 8, 3:00 pm — 4:30 pm
P10-1 A New Low-Delay Codec for Two-Way High-Quality Audio Communication—Aníbal Ferreira, University of Porto - Porto, Portugal and ATC Labs, Chatham, NJ, USA; Deepen Sinha, ATC Labs - Chatham, NJ, USA
High-quality audio bit-rate reduction systems are widely used in many application areas involving audio broadcast, streaming, and download services. With the advent of 3G mobile and wireless communication networks, there is a clear opportunity for new multimedia services, notably those relying on two-way high-quality audio communication. In this paper we describe a new source/perceptual audio coder that features low-delay, intrinsic error robustness, and high subjective audio quality at competitive compression ratios. The structure of the audio coder is described and an emphasis is given on its innovative approaches to semantic signal segmentation and decomposition, independent coding of sinusoidal and noise components, and bandwidth extension using accurate spectral replacement. A few test results are presented that illustrate the operation and performance of the new coder. Audio demonstations are available at http://www.atc-labs.com/acc/.
Convention Paper 6572 (Purchase now)
P10-2 Compensation of Nonlinearities of Horn Loudspeakers—Delphine Bard, Mario Rossi, Ecole Polytechnique Federale de Lausanne - Lausanne, Switzerland; Mauro Del Nobile, Swissphonics - Peseux, Switzerland
This paper presents a compensation method of nonlinearities of horn loudspeakers. It is possible to compensate the nonlinearity’s effects of electroacoustic devices by applying the inverse nonlinearity upstream. The method is based on the measurements of the nonlinearity by Volterra series using multitone excitations. Once the Volterra kernels have been determined, we proceed by computing the inverse Volterra kernels, both in magnitude and phase. The method was implemented and validated in non-real time and real time (DSP implementation). To validate the nonlinearities compensation a comparison between total harmonic distortion measurements with and without compensation has been done.
Convention Paper 6573 (Purchase now)
P10-3 Diaphragm Parameters and Radiation Characteristics of Multilayer Piezoelectric Ceramic Loudspeakers—Jun Fujii, Juro Ohga, Shibaura Institute of Technology - Mitato-ku,Tokyo, Japan; Norikazu Sashida, Ashida Sound Co. - Shinagawa-ku, Tokyo, Japan; Ikuo Oohira, Taiyo Yuden Co., Ltd. - Hauna-machi., Gunma., Japan
This paper presents diaphragm parameters and analysis of radiation characteristics for small size loudspeaker by a multilayer piezoelectric ceramic bimorph diaphragm. The multilayer ceramic wafer is suitable for battery-operated mobile phones because of its lower electrical impedance nature. Three diaphragm parameter measuring methods are compared to develop the optimum measurement of diaphragm parameters. Then, output sound pressure frequency characteristics of a loudspeaker model with actual acoustical loads are analyzed.
Convention Paper 6574 (Purchase now)
P10-4 A Study on Lumped Elements Model and Thermal Effects of Eddy Currents in Loudspeakers—Ning Wu, Yong Shen, Xiaobing Xu, Nanjing University - Nanjing, China
A frequency-divided thermal model is developed to study the heat arising from the eddy currents in electro-dynamic loudspeakers. Using pure tone as the test signal, the steady state temperature of the voice coil is measured point by point in a high frequency range. The results illuminate the contrast of all the existing lumped electrical-models of eddy currents and show distinctly which is better. Also, a set of innovative thermal expressions considering different circumstances are deduced from the simplified thermal model. With these expressions and the measurement data, all the thermal elements’ value in the model can be obtained. Arbitrary temperature-rising course can then be predicted easily if several necessary parameters are given.
Convention Paper 6575 (Purchase now)
P10-5 Spatial Audio Coding System Based on Virtual Source Location Information—Jeongil Seo, Inseon Jang, Kyeongok Kang, Electronics and Telecommunications Research Institute (ETRI) - Daejon, Korea
Spatial audio coding (SAC) is a process to represent multichannel audio signals as down-mixed mono or stereo signals with spatial cues. The main strength of SAC is the significant bit-rate reduction while maintaining the perceptual sound quality. Binaural cue coding (BCC) has been introduced and now becomes an important scheme for multichannel SAC both in the sense of audio coding and the standardization issue in the MPEG. However, interchannel level difference (ICLD), one of the essential spatial cues for SAC, has a limitation that the quantized ICLD for transmission may lead to the sound quality degradation of a decoded signal. In this paper we propose virtual source location information (VSLI), which is an angle representing geometric spatial information between channels on playback layout, instead of the ICLD, and also a VSLI-based SAC system. Since a human being cannot easily distinguish the variation of the spatial angle within the three degree distortion, the spatial angle, hence the VSLI, can be approximated discretely with the three degree resolution while maintaining the perceptual quality of output signals. The objective and subjective assessment results of our proposed system confirm superior performance to the ICLD-based SAC system.
Convention Paper 6576 (Purchase now)
P10-6 An Ultra High Performance DAC with Controlled Time-Domain Response—Paul Lesso, Anthony Magrath, Wolfson Microelectronics - Edinburgh, Scotland, UK
This paper describes the design of an ultra-high performance stereo digital-to-analog converter (DAC) employing advanced digital filtering techniques. Recently there has been a renewed interest in the time-domain properties of digital filters used for interpolation and decimation. Linear phase FIR filters, which have proliferated digital filter design for the last two decades, have the undesirable properties of pre-ringing and high group delay. Conversely, minimum phase filters, which offer lower levels of pre-ringing, do not have a uniform phase response. This paper describes the trade-offs in the design of filters with controlled pre-ringing, coupled with desirable phase and magnitude characteristics. The paper also describes architectural choices in the implementation of the DAC signal processing chain, required to achieve commensurate analog performance.
Convention Paper 6577 (Purchase now)
P10-7 Understanding the Effects of AES-17 When Evaluating 192-kHz Converter Performance—Richard Kulavik, Larry Gaddy, AKM Semiconductors, Inc. - San Jose, CA, USA
This paper will cover hidden performance issues in 192-kHz converters that are evaluated using AES-17. AES-17 is the “AES Standard method for digital audio engineering—Measurement of digital audio equipment.” This standard calls out how to test most digital audio equipment, and it outlines the standard test procedures and methods in testing of audio. It is important to understand that these methods allow for several important issues to be hidden in the real spectrum of the converter. These hidden elements will be addressed in detail. Specifically, sections 9.1 and 9.3 of the AES standard will be examined.
Convention Paper 6578 (Purchase now)
