AES San Francisco 2012
Poster Session P7
P7 - Amplifiers, Transducers, and Equipment
Friday, October 26, 3:00 pm — 4:30 pm (Foyer)
P7-1 Evaluation of trr Distorting Effects Reduction in DCI-NPC Multilevel Power Amplifiers by Using SiC Diodes and MOSFET Technologies—Vicent Sala, UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain; Tomas Resano, Jr., UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain; MCIA Research Center; Jose Luis Romeral, UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain; Jose Manuel Moreno, UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain
In the last decade, the Power Amplifier applications have used multilevel diode-clamped-inverter or neutral-point-clamped (DCI-NPC) topologies to present very low distortion at high power. In these applications a lot of research has been done in order to reduce the sources of distortion in the DCI-NPC topologies. One of the most important sources of distortion, and less studied, is the reverse recovery time (trr) of the clamp diodes and MOSFET parasitic diodes. Today, with the emergence of Silicon Carbide (SiC) technologies, these sources of distortion are minimized. This paper presents a comparative study and evaluation of the distortion generated by different combinations of diodes and MOSFETs with Si and SiC technologies in a DCI-NPC multilevel Power Amplifier in order to reduce the distortions generated by the non-idealities of the semiconductor devices.
Convention Paper 8720 (Purchase now)
P7-2 New Strategy to Minimize Dead-Time Distortion in DCI-NPC Power Amplifiers Using COE-Error Injection—Tomas Resano, Jr., UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain; MCIA Research Center; Vicent Sala, UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain; Jose Luis Romeral, UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain; Jose Manuel Moreno, UPC-Universitat Politecnica de Catalunya - Terrassa, Catalunya, Spain
The DCI-NPC topology has become one of the best options to optimize energy efficiency in the world of high power and high quality amplifiers. This can use an analog PWM modulator that is sensitive to generate distortion or error, mainly for two reasons: Carriers Amplitude Error (CAE) and Carriers Offset Error (COE). Other main error and distortion sources in the system is the Dead-Time (td). This is necessary to guarantee the proper operation of the power amplifier stage so that errors and distortions originated by it are unavoidable. This work proposes a negative COE generation to minimize the distortion effects of td. Simulation and experimental results validates this strategy.
Convention Paper 8721 (Purchase now)
P7-3 Further Testing and Newer Methods in Evaluating Amplifiers for Induced Phase and Frequency Modulation via Tones, Amplitude Modulated Signals, and Pulsed Waveforms—Ronald Quan, Ron Quan Designs - Cupertino, CA, USA
This paper will present further investigations from AES Convention Paper 8194 that studied induced FM distortions in audio amplifiers. Amplitude modulated (AM) signals are used for investigating frequency shifts of the AM carrier signal with different modulation frequencies. A square-wave and sine-wave TIM test signal is used to evaluate FM distortions at the fundamental frequency and harmonics of the square-wave. Newer amplifiers are tested for FM distortion with a large level low frequency signal inducing FM distortion on a small level high frequency signal. In particular, amplifiers with low and higher open loop bandwidths are tested for differential phase and FM distortion as the frequency of the large level signal is increased from 1 KHz to 2 KHz.
Convention Paper 8722 (Purchase now)
P7-4 Coupling Lumped and Boundary Element Methods Using Superposition—Joerg Panzer, R&D Team - Salgen, Germany
Both, the Lumped and the Boundary Element Method are powerful tools for simulating electroacoustic systems. Each one can have its preferred domain of application within one system to be modeled. For example the Lumped Element Method is practical for electronics, simple mechanics, and internal acoustics. The Boundary Element Method on the other hand enfolds its strength on acoustic-field calculations, such as diffraction, reflection, and radiation impedance problems. Coupling both methods allows to investigate the total system. This paper describes a method for fully coupling of the rigid body mode of the Lumped to the Boundary Element Method with the help of radiation self- and mutual radiation impedance components using the superposition principle. By this, the coupling approach features the convenient property of a high degree of independence of both domains. For example, one can modify parameters and even, to some extent, change the structure of the lumped-element network without the necessity to resolve the boundary element system. This paper gives the mathematical derivation and a demonstration-example, which compares calculation results versus measurement. In this example electronics and mechanics of the three involved loudspeakers are modeled with the help of the lumped element method. Waveguide, enclosure and radiation is modeled with the boundary element method.
Convention Paper 8723 (Purchase now)
P7-5 Study of the Interaction between Radiating Systems in a Coaxial Loudspeaker—Alejandro Espi, Acústica Beyma - Valencia, Spain; William A. Cárdenas, Sr., University of Alicante - Alicante, Spain; Jose Martinez, Acustica Beyma S.L. - Moncada (Valencia), Spain; Jaime Ramis, University of Alicante - Alicante, Spain; Jesus Carbajo, University of Alicante - Alicante, Spain
In this work the procedure followed to study the interaction between the mid and high frequency radiating systems of a coaxial loudspeaker is explained. For this purpose a numerical Finite Element model was implemented. In order to fit the model, an experimental prototype was built and a set of experimental measurements, electrical impedance, and pressure frequency response in an anechoic plane wave tube among these, were carried out. So as to take into account the displacement dependent nonlinearities, a different input voltage parametric analysis was performed and internal acoustic impedance was computed numerically in the frequency domain for specific phase plug geometries. Through inversely transforming to a time differential equation scheme, a lumped element equivalent circuit to evaluate the mutual acoustic load effect present in this type of acoustic coupled systems was obtained. Additionally, the crossover frequency range was analyzed using the Near Field Acoustic Holography technique.
Convention Paper 8724 (Purchase now)
P7-6 Flexible Acoustic Transducer from Dielectric-Compound Elastomer Film—Takehiro Sugimoto, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Tokyo Institute of Technology - Midori-ku, Yokohama, Japan; Kazuho Ono, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Akio Ando, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Hiroyuki Okubo, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Kentaro Nakamura, Tokyo Institute of Technology - Midori-ku, Yokohama, Japan
To increase the sound pressure level of a flexible acoustic transducer from a dielectric elastomer film, this paper proposes compounding various kinds of dielectrics into a polyurethane elastomer, which is the base material of the transducer. The studied dielectric elastomer film utilizes a change in side length derived from the electrostriction for sound generation. The proposed method was conceived from the fact that the amount of dimensional change depends on the relative dielectric constant of the elastomer. Acoustical measurements demonstrated that the proposed method was effective because the sound pressure level increased by 6 dB at the maximum.
Convention Paper 8725 (Purchase now)
P7-7 A Digitally Driven Speaker System Using Direct Digital Spread Spectrum Technology to Reduce EMI Noise—Masayuki Yashiro, Hosei University - Koganei, Tokyo, Japan; Mitsuhiro Iwaide, Hosei University - Koganei, Tokyo, Japan; Akira Yasuda, Hosei University - Koganei, Tokyo, Japan; Michitaka Yoshino, Hosei University - Koganei, Tokyo, Japan; Kazuyki Yokota, Hosei University - Koganei, Tokyo, Japan; Yugo Moriyasu, Hosei University - Koganei, Tokyo, Japan; Kenji Sakuda, Hosei University - Koganei, Tokyo, Japan; Fumiaki Nakashima, Hosei University - Koganei, Tokyo, Japan
In this paper a novel digital direct-driven speaker for reducing electromagnetic interference incorporating a spread spectrum clock generator is proposed. The driving signal of a loudspeaker, which has a large spectrum at specific frequency, interferes with nearby equipment because the driving signal emits electromagnetic waves. The proposed method changes two clock frequencies according to the clock selection signal generated by a pseudo-noise circuit. The noise performance deterioration caused by the clock frequency switching can be reduced by the proposed modified delta-sigma modulator, which changes coefficients of the DSM according to the width of the clock period. The proposed method can reduce out-of-band noise by 10 dB compared to the conventional method.
Convention Paper 8726 (Purchase now)
P7-8 Automatic Speaker Delay Adjustment System Using Wireless Audio Capability of ZigBee Networks—Jaeho Choi, Seoul National University - Seoul, Korea; Myoung woo Nam, Seoul National University - Seoul, Korea; Kyogu Lee, Seoul National University - Seoul, Korea
IEEE 802.15.4 (ZigBee) standard is a low data rate, low power consumption, low cost, flexible network system that uses wireless networking protocol for automation and remote control applications. This paper applied these characteristics on the wireless speaker delay compensation system in a large venue (over 500-seat hall). Traditionally delay adjustment has been manually done by sound engineers, but our suggested system will be able to analyze delayed-sound of front speaker to rear speaker automatically and apply appropriate delay time to rear speakers. This paper investigates the feasibility of adjusting the wireless speaker delay over the above-mentioned ZigBee network. We present an implementation of a ZigBee audio transmision and LBS (Location-Based Service) application that allows to calculation a speaker delay time.
Convention Paper 8727 (Purchase now)
P7-9 A Second-Order Soundfield Microphone with Improved Polar Pattern Shape—Eric M. Benjamin, Surround Research - Pacifica, CA, USA
The soundfield microphone is a compact tetrahedral array of four figure-of-eight microphones yielding four coincident virtual microphones; one omnidirectional and three orthogonal pressure gradient microphones. As described by Gerzon, above a limiting frequency approximated by fc = pc/r, the virtual microphones become progressively contaminated by higher-order spherical harmonics. To improve the high-frequency performance, either the array size must be substantially reduced or a new array geometry must be found. In the present work an array having nominally octahedral geometry is described. It samples the spherical harmonics in a natural way and yields horizontal virtual microphones up to second order having excellent horizontal polar patterns up to 20 kHz.
Convention Paper 8728 (Purchase now)
P7-10 Period Deviation Tolerance Templates: A Novel Approach to Evaluation and Specification of Self-Synchronizing Audio Converters—Francis Legray, Dolphin Integration - Meylan, France; Thierry Heeb, Digimath - Sainte-Croix, Switzerland; SUPSI, ICIMSI - Manno, Switzerland; Sebastien Genevey, Dolphin Integration - Meylan, France; Hugo Kuo, Dolphin Integration - Meylan, France
Self-synchronizing converters represent an elegant and cost effective solution for audio functionality integration into SoC (System-on-Chip) as they integrate both conversion and clock synchronization functionalities. Audio performance of such converters is, however, very dependent on the jitter rejection capabilities of the synchronization system. A methodology based on two period deviation tolerance templates is described for evaluating such synchronization solutions, prior to any silicon measurements. It is also a unique way for specifying expected performance of a synchronization system in the presence of jitter on the audio interface. The proposed methodology is applied to a self-synchronizing audio converter and its advantages are illustrated by both simulation and measurement results.
Convention Paper 8729 (Purchase now)
P7-11 Loudspeaker Localization Based on Audio Watermarking—Florian Kolbeck, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Giovanni Del Galdo, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Iwona Sobieraj, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Tobias Bliem, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Localizing the positions of loudspeakers can be useful in a variety of applications, above all the calibration of a home theater setup. For this aim, several existing approaches employ a microphone array and specifically designed signals to be played back by the loudspeakers, such as sine sweeps or maximum length sequences. While these systems achieve good localization accuracy, they are unsuitable for those applications in which the end-user should not be made aware that the localization is taking place. This contribution proposes a system that fulfills these requirements by employing an inaudible watermark to carry out the localization. The watermark is specifically designed to work in reverberant environments. Results from realistic simulations confirm the practicability of the proposed system.
Convention Paper 8730 (Purchase now)
