AES New York 2009
Poster Session P11

P11 - Sound in Real Spaces

Saturday, October 10, 3:30 pm — 5:00 pm
P11-1 Acoustics of National Parks and Historic Sites: The 8,760 Hour MP3 File—Robert Maher, Montana State University - Bozeman, MT, USA
According to current U.S. National Park Service (NPS) management policies, the natural soundscape of parks and historic sites is a protected resource just like the ecosystems, landscapes, and historic artifacts for which the parks were formed. While several NPS sites have been studied extensively for noise intrusions by tour aircraft and mechanized recreation, most parks and historic sites do not yet have an acoustic baseline for management purposes. A recent initiative of the NPS Natural Sounds Office is to obtain continuous audio recordings of specific sites for one entire year. This paper explores the engineering and scientific issues associated with obtaining, archiving, and cataloging an 8,760 hour long audio recording for Grant-Kohrs Ranch National Historic Site.
Convention Paper 7893 (Purchase now)

P11-2 Improved Speech Dereverberation Method Using the Kurtosis-Maximization with the Voiced/Unvoiced/Silence Classification—Jae-woong Jeong, Se-Woon Jeon, Yonsei University - Seoul, Korea; Young-cheol Park, Yonsei University - Wonju, Korea; Seok-Pil Lee, Korea Electronics Technology Institute (KETI) - Sungnam, Korea; Dae-hee Youn, Yonsei University - Seoul, Korea
In this paper we present a new speech dereverberation method using the kurtosis-maximization based on the voiced/unvoiced/silence (V/UV/S) classification. Since kurtosis of the UV/S sections are much smaller than V sections, adaptation of the dereverberation filter using these sections often results in slow and nonrobust convergence, and, in turn, poor dereverberation. The proposed algorithm controls adaptation of the dereverberation filter using the results of V/UV/S classification, together with kurtosis measure of the input speech. For the selective control of adaptation, both hard decision and voice likelihood measure based on various features together with kurtosis were tried, and then, the step-size of the adaptive algorithm was varied according to various control strategies. The proposed algorithm provides better and more robust dereverberation performance than the conventional algorithm, which was confirmed through the experiments.
Convention Paper 7894 (Purchase now)

P11-3 A Survey of Broadcast Television Perceived Relative Audio Levels—Chris Hanna, Matthew Easley, THAT Corporation - Milford, MA, USA
Perceived television volume levels can vary dramatically as audio changes both within a given broadcast channel and between broadcast channels. This paper surveys the broadcast audio levels in two large metropolitan areas (Atlanta and Boston). Both analog and digital broadcasts are monitored from cable and satellite providers. Two-channel perceived loudness is measured utilizing the ITU-R Rec. BS.1770 loudness meter standard. Statistical data is presented showing the severity and nature of the perceived loudness changes. Finally, dynamic volume control technology is applied to the most severe recordings for perceived loudness comparisons.
Convention Paper 7896 (Purchase now)

P11-4 Optimizing the Re-enforcement Effect of Early Reflections on Aspects of Live Musical Performance Using the Image Source Model—Michael Terrell, Joshua Reiss, Queen Mary University of London - London, UK
The image source method is used to identify early reflections which have a re-enforcement effect on the sound traveling within an enclosure. The distribution of absorptive material within the enclosure is optimized to produce the desired re-enforcement effect. This is applied to a monitor mix and a feedback prevention case study. In the former it is shown that the acoustic path gain of the vocals can be increased relative to the acoustic path gain of the other instruments. In the latter it is shown that the acoustic path from loudspeaker to microphone can be manipulated to increase the perceived signal level before the onset of acoustic feedback.
Convention Paper 7897 (Purchase now)

P11-5 The Influence of the Rendering Architecture on the Subjective Performance of Blind Source Separation Algorithms—Thorsten Kastner, University of Erlangen-Nuremberg - Erlangen, Germany, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany
Blind Source Separation algorithms often include a time/frequency (t/f) decomposition / filterbank as an important part allowing for frequency selective separation of the input signal. In order to investigate the importance of the t/f processing architecture for the achieved subjective audio quality, a set of blindly separated audio signals were taken from the Stereo Audio Source Separation Campaign (SASSEC) 2007 and rated in a MUSHRA listening test. The test furthermore included material that was generated by using the separated signals to drive an enhanced time/frequency rendering architecture, as it is offered by MPEG Spatial Audio Object Coding (SAOC). In this way, the same basic separation core algorithm was applied together with different t/f rendering architectures. The listening test reveals an improved subjective quality for the SAOC post-processed items.
Convention Paper 7898 (Purchase now)

P11-6 Real-Time Implementation of Robust PEM-AFROW-Based Solutions for Acoustic Feedback Control—Simone Cifani, Rudy Rotili, Emanuele Principi, Stefano Squartini, Francesco Piazza, Università Politecnica delle Marche - Ancona, Italy
Acoustic feedback is a longstanding problem in the audio processing field, occurring whenever sound is captured and reproduced in the same environment. Different control strategies have been proposed over the years, among which a feedback cancellation technique based on the prediction error method (PEM) has revealed to be performing on purpose. Recent studies have shown that the integration of a suppression or a noise reduction filter in the system loop might be beneficial from different perspectives. In this paper a real-time implementation of the aforementioned algorithm is presented, which exploits the partitioned-block frequency-domain (PBFD) technique to allow the system to work also with long acoustic paths. NU-Tech software platform has been used on purpose for real-time simulations, performed over synthetic and real acoustic conditions.
Convention Paper 7899 (Purchase now)

P11-7 Perception-Based Audio Signal Mixing in Automotive Environments—Wolfgang Hess, Harman/Becker Automotive Systems - Karlsbad-Ittersbach, Germany
Information and announcement presentation in noisy environments such as vehicles requires dynamic adjustment of signals for optimal information audibility and speech intelligibility. Not only variant ambient noises, but, in particular, their combination with today’s vehicle infotainment systems capability to reproduce a variety of entertainment signal sources, make information presentation difficult. Most different input level ranges as well as a variety of compressions ratios of audio signals have to be considered. A further challenge is the dynamic, loudness-dependent binaural intelligibility level difference of the human auditory system. The algorithm presented in this paper solves these issues described here by dynamically mixing information and announcement signals to entertainment signals. Entertainment signals are attenuated as little as possible, and information or announcement signals are added in loudness as demanded. As a result, optimal announcement intelligibility and information audibility is achieved.
Convention Paper 7900 (Purchase now)

P11-8 Visualization and Analysis Tools for Low Frequency Propagation in a Generalized 3-D Acoustic Space—Adam J. Hill, Malcolm O. J. Hawksford, University of Essex - Colchester, Essex, UK
A toolbox is described that enables 3-D animated visualization and analysis of low-frequency wave propagation within a generalized acoustic environment. The core computation exploits a Finite-Difference Time-Domain (FDTD) algorithm selected because of its known low frequency accuracy. Multiple sources can be configured and analyses performed at user-selected measurement locations. Arbitrary excitation sequences enable virtual measurements embracing both time-domain and spatio-frequency domain analysis. Examples are presented for a variety of low-frequency loudspeaker placements and room geometries to illustrate the versatility of the toolbox as an acoustics design aid.
Convention Paper 7901 (Purchase now)