AES Warsaw 2015
Poster Session P16
P16 - (Poster) Applications in Audio
Sunday, May 10, 10:00 — 12:00 (Foyer)
P16-1 Dubbing Studio for 22.2 Multichannel Sound System in NHK Broadcasting Center—Ikuko Sawaya, Science & Technology Research Laboratories, Japan Broadcasting Corp (NHK). - Setagaya, Tokyo, Japan; Kengo Sasaki, Japan Broadcasting Corporation - Shibuya, Tokyo, Japan; Shinji Mikami, Japan Broadcasting Corporation - Shibuya, Tokyo, Japan; Hiroyuki Okubo, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Kazuho Ono, NHK Engineering System Inc. - Setagaya-ku, Tokyo, Japan
8K Super Hi-Vision is planned to be on test broadcasting in 2016 and to launch a full broadcasting service in 2018 in Japan. NHK has developed a program production system for 22.2 multichannel sound for 8K Super Hi-Vision. As part of the development NHK completed the construction of a 22.2 ch dubbing studio in the NHK Broadcasting Center in July 2014. This is the first 22.2 ch dubbing studio in the production field in the world with a loudspeaker configuration that meets the standard Recommendation ITU-R BS.2051. In this paper we discuss the 22.2 ch production system, including its sound mixing system, loudspeaker system for monitoring, and perforated screen for 8K resolution, as well as the room design and the characteristics of the room acoustics in the studio.
Convention Paper 9327 (Purchase now)
P16-2 A Floor Acoustic Sensor for Fall Classification—Emanuele Principi, Università Politecnica delle Marche - Ancona, Italy; Paolo Olivetti, Scientific Direction, Italian National Institute of Health and Science on Aging (INRCA) - Ancona, Italy; Stefano Squartini, Università Politecnica delle Marche - Ancona, Italy; Roberto Bonfigli, Universita Politecnica delle Marche - Ancona, Italy; Francesco Piazza, Universitá Politecnica della Marche - Ancona (AN), Italy
The interest in assistive technologies for supporting people at home is constantly increasing, both in academia and industry. In this context the authors propose a fall classification system based on an innovative acoustic sensor that operates similarly to stethoscopes and captures the acoustic waves transmitted through the floor. The sensor is designed to minimize the impact of aerial sounds in recordings, thus allowing a more focused acoustic description of fall events. In this preliminary work, the audio signals acquired by means of the sensor are processed by a fall recognition algorithm based on Mel-Frequency Cepstral Coefficients, Supervectors, and Support Vector Machines to discriminate among different types of fall events. The performance of the algorithm has been evaluated against a specific audio corpus comprising falls of persons and of common objects. The results show the effectiveness of the approach.
Convention Paper 9329 (Purchase now)
P16-3 Active Field Control in the Teatr Wielki—Opera Narodowa—Takayuki Watanabe, Yamaha Corp. - Hamamatsu, Shizuoka, Japan; Hideo Miyazaki, Yamaha Corp. - Hamamatsu, Shizuoka, Japan; Shinichi Sawara, Yamaha Corp. - Hamamatsu, Shizuoka, Japan; Masahiro Ikeda, Yamaha Corporation - Hamamatsu, Shizuoka, Japan; Ron Bakker, Yamaha Commercial Audio Systems Europe - Rellingen, Germany
This opera house of 1,828 seats boasts one of Europe's largest stages and is highly reputed for its repertoire and acoustics. However, it presented a number of issues including poor communication between the singers and the orchestra pit, insufficient loudness of the upstage singers for the audience, a lack of reverberation when the house was occupied, and insufficient loudness at the seats under the balconies. For these reasons Active Field Control System (AFC) was adopted as a means to improve the acoustics while preserving the historic architecture of the opera house. This paper presents an overview of that system and the benefits achieved by its introduction.
Convention Paper 9330 (Purchase now)
P16-4 An Environment for Submillisecond-Latency Audio and Sensor Processing on BeagleBone Black—Andrew McPherson, Queen Mary University of London - London, UK; Victor Zappi, University of British Columbia - Vancouver, BC, Canada
This paper presents a new environment for ultra-low-latency processing of audio and sensor data on embedded hardware. The platform, which is targeted at digital musical instruments and audio effects, is based on the low-cost BeagleBone Black single-board computer. A custom expansion board features stereo audio and 8 channels each of 16-bit ADC and 16-bit DAC for sensors and actuators. In contrast to typical embedded Linux approaches, the platform uses the Xenomai real-time kernel extensions to achieve latency as low as 80 microseconds, making the platform suitable for the most demanding of low-latency audio tasks. The paper presents the hardware, software, evaluation, and applications of the system.
Convention Paper 9331 (Purchase now)
P16-5 Commonwealth Games 2014 Host Broadcaster Training Initiative–A Game Changer?—Patrick Quinn, Glasgow Caledonian University - Glasgow, Scotland, UK; David Moore, Glasgow Caledonian University - Glasgow, Lanarkshire, UK
Glasgow Commonwealth Games 2014 provided an ideal platform for over 200 students to gain work experience in sports broadcasting as part of the Host Broadcaster Training Initiative. Organized by SVGTV and Creative Loop with the intention of attracting students into this growing area of broadcasting, the successful initiative has encouraged many of the students involved, including Audio Technology students from Glasgow Caledonian University, to consider and subsequently pursue careers in broadcasting. In addition as a legacy from the initiative a new course is planned at Glasgow Caledonian University in Broadcasting Technology.
Convention Paper 9332 (Purchase now)
P16-6 Influence of Noise on the Effectiveness of Speaker Identification in the Acoustics of Crime—Tomasz Smutnicki, Wroclaw University of Technology - Wroclaw, Poland; Stefan Brachmanski, Wroclaw University of Technology - Wroclaw, Poland
One of the main elements of the research in acoustics of crime is to compare the evidential recording with the comparative adequate pattern. Unfortunately, the evidential recording usually has poor quality and contains relatively high level of noise, which results from the way of its acquiring, namely eavesdropping or record of automatic monitoring. The signal quality and the noise to signal ratio have an impact on the value of the extracted voice metrics. In this paper we analyze factors that may have an impact on formants value in the human voice. Based on Six Sigma methodology we also designed and performed an experiment that allowed us to determine the extent in which various factors influence on the resulting parameters.
Convention Paper 9333 (Purchase now)
P16-7 Acoustic Profile of Identified Speaker in Forensics—Krystian Kapala, Wroclaw University of Technology - Wroclaw, Poland; Stefan Brachmanski, Wroclaw University of Technology - Wroclaw, Poland
Speaker identification is deemed to be one of the basic tasks in audio forensics. Delivering a categorical opinion is often difficult due to insufficient quality of the recorded material, simulation or modulation of speaker’s voice. Hence, a wide-ranging approach to the identification process is used, including both subjective and objective methods. With their help, it becomes possible to obtain a broad spectrum of speech characteristics ranging from low-level features relating to physical construction of the vocal tract to advanced ones concerning various ways of expressing oneself and articulation, acquired during socialization process. This article describes an experiment undertaken to create acoustic profiles of a chosen group of speakers based on the features mentioned above.
Convention Paper 9334 (Purchase now)
P16-8 An Implementation Of Beamforming Algorithm On FPGA Platform with Digital Microphone Array—Iva Salom, Institute Mihajlo Pupin, University of Belgrade - Belgrade, Serbia; Vladimir Celebic, Institute Mihailo Pupin, University of Belgrade - Belgrade, Serbia; Milan Milanovic, Institute Mihailo Pupin, University of Belgrade - Belgrade, Serbia; Dejan Todorovic, Dirigent Acoustics Ltd. - Belgrade, Serbia; Jurij Prezelj, University of Ljubljana - Ljubljana, Slovenia
The goal of the project described in this paper was to design an acoustic system for localization of the dominant noise source by implementation of the conventional delay-and-sum beamforming algorithm on FPGA platform with a sound receiver system based on digital MEMS microphone array. The system consists of a platform for acoustic signal acquisition and data processing (microphone array, interface, and central block), and a platform for monitoring and control (a computer with a user application). Such configuration provides the execution of the beamforming algorithm in real time. Additionally, FPGAs are bringing many benefits in terms of safety, reliability, rapidity, and power consumption. The platform was tested and verified with various microphone array configurations and results are presented in the paper.
Convention Paper 9335 (Purchase now)
P16-9 Measuring and Analyzing Audio Levels in Film, Commercials, and Movie Trailers Using Leq(A) Values and the LUFS Loudness Model—Bozena Kostek, Gdansk University of Technology - Gdansk, Poland; Audio Acoustics Lab.; Kamila Milarska, Gdansk University of Technology - Gdansk, Poland; Aleksander Zakrzewski, Gdansk University of Technology - Gdansk, Poland
The purpose of this paper is to describe the measurement of loudness levels in movies, movie trailers, and commercials displayed before feature films at movie theaters. In the initial section, the paper discusses the issues related to measurement of loudness levels, provides recommendations regarding permissible loudness levels during movie screenings, and mentions the applied units of measurement. The following section of the paper describes the actual measurements, measuring equipment, as well as analysis of the results of the measurements. The summary provides conclusions about the measured loudness levels at movie theaters, for DVD and Blu-ray discs, and for YouTube videos.
Convention Paper 9336 (Purchase now)
P16-10 Arbitrary Trajectory Estimation of a Moving Acoustic Source—Sai Gunaranjan Pelluri, Indian Institute of Science - Bangalore, India; Thippur V. Sreenivas, Indian Institute of Science - Bangalore, India
The state-of-the-art passive methods for estimating the trajectory of a moving acoustic source involve computing the cross-correlation function either directly or indirectly (as in the case of the Beam Forming Approach) between pairs of microphones. Also there have been several Active Acoustic techniques such as SONAR, RADAR, etc., which have been used to estimate the source parameters such as velocity, trajectory, etc. They involve pinging the source with a known signal. In this paper, given the fact that the moving source itself generates a signal, we propose a technique by which we estimate the source trajectory using only the signal captured at the receiver thereby avoiding the need to ping the source and without computing the cross-correlation function.
Convention Paper 9266 (Purchase now)
