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Last Updated: 20070417, meiP12 - Microphones and Loudspeakers & Audio in Computers (Games, Internet, Desktop Computer Audio)
Sunday, May 6, 13:00 — 14:30
P12-1 The Relation between Active Radiating Factor and Pressure Responses of Loudspeaker Line Arrays—Yong Shen, Dayi Ou, Kang An, Nanjing University - Nanjing, China
Active Radiating Factor (ARF) is an important parameter to analyze the loudspeaker line array when considering the gaps between each of two radiating transducers. The relation between ARF of the loudspeaker line array and the differential chart of its pressure responses in two distances (PRD) is analyzed. Some valuable conclusions about ARF and PRD are found. A method to estimate ARF by measuring pressure responses comes out.
Convention Paper 7058 (Purchase now)
P12-2 Alternative Encoding Techniques for Digital Loudspeaker Arrays—Fotios Kontomichos, Nicolas–Alexander Tatlas, John Mourjopoulos, University of Patras - Patras, Greece
Recent developments in digital loudspeakers have resulted in the introduction of digital transducer arrays (DTA). In most implementations, DTA loudspeakers are driven by PCM encoded audio signals, usually resampled and requantized to an appropriate number of bits, in accordance to the number of the transducers constituting the DTA topology. However, given that DTAs generally increase harmonic distortion, especially for off-axis listening positions, optimization in signal encoding and bit-to-transducer assignment, is necessary. Here, a number of novel, alternative strategies are examined, concerning the input signal encoding via PCM-to-PWM conversion, as well as techniques for bit-assignment on the transducers of a DTA. These tests are supported by simulation results and comparisons, for different operating parameters.
Convention Paper 7059 (Purchase now)
P12-3 Online Identification of Linear Loudspeaker Parameters—Bo Rohde Pedersen, Aalborg University - Esbjerg, Denmark; Per Rubak, Aalborg University - Aalborg, Denmark
Feed forward nonlinear error correction of loudspeakers can improve sound quality. For creating an efficient feed forward strategy identification of the loudspeaker parameters is needed. The strategy of the compensator is that the nonlinear behavior of the loudspeakers has relatively small drift and only the linear loudspeaker parameters must be identified. In music systems this can be done with online transducer-less system identification using the voice coil current as feedback from the loudspeaker (plant). This is investigated in a simulation study for finding useful system identification algorithms. Two different identification techniques (ARMA and FIR) are compared. The stability of the nonlinearities is tested in a measurement series.
Convention Paper 7060 (Purchase now)
P12-4 Finite Element Analysis of Near Field Beam Forming in Safety Relevant Work Spaces—Roman Beigelbeck, Austrian Academy of Sciences - Wiener Neustadt, Austria; Heinrich Pichler, Consultant - Vienna, Austria
Due to their unique features, loudspeaker arrays are an interesting alternative to standard loudspeaker setups or headphone-based solutions in safety relevant workspaces such as air traffic control rooms. Consequentially, near field beam forming in small spaces plays an important role for this field of application. In this paper the sound design based on a set of loudspeaker arrays featuring their interaction with a typical air traffic control room infrastructure is investigated by means of finite element modeling. Guided by these results, optimized array parameters can be determined. Representative three-dimensional near field directional diagrams in front of the arrays are shown to visualize the sound field in different cases. Finally, these theoretical values are compared with practical results.
Convention Paper 7061 (Purchase now)
P12-5 Creating Directed Microphones from Undirected Microphones—Emil Milanov, Elena Milanova, Acoustical Engineers - Sofia, Bulgaria
In this paper we examine the possibility of creating directed microphones from undirected microphones. The result is achieved only by using acoustical elements and is valid for all types of microphones, regardless of their way of work (electro-dynamical, condenser, optical, electro-mechanical, etc.). By using alternations of the membrane usage, a force is obtained, which is equivalent to the effect of simultaneously operating undirected and bidirected (eight) microphones. The result is a microphone with a space characteristic equivalent to the Pascal curve (i.e., directed microphone with the traditional shapes of the space characteristic—cardioid, super cardioid and hyper cardioid). The shape of the space characteristic curve is near to theoretical and does not depend on the acoustic elements of the microphone. The microphone is directed, but does not have a proximity effect.
Convention Paper 7062 (Purchase now)
P12-6 Transducer with the Direct D/A Conversion Using the Optoacoustic Principle—Libor Husník, Czech Technical University in Prague - Prague, Czech Republic
Transducers with the direct D/A conversion, sometimes called digital transducers, either loudspeakers or earphones, are searching their ways into being. There have been several attempts to design such devices, but none of them left research laboratories and made its way to commercial use as yet. Most of them use “classical” electroacoustic transduction principles, i.e., electrodynamic or electrostatic. In this paper the possibility to use optoacoustic transduction principles is explored. First, the principles of physical phenomena used in this transducer are revised. Then, some construction details in light of their usage in the digital earphone are described.
Convention Paper 7064 (Purchase now)
P12-7 Demystifying the Measurement of Impulse Response in Condenser Microphones—Part I—Christian Langen, Schoeps Mikrofone - Karlsruhe, Germany
Good impulse response is an important reason for preferring condenser microphones in audio applications that require high quality. However, it is difficult to characterize the impulse response of a microphone precisely. We cannot create an acoustic impulse that approximates the Dirac delta function closely enough that a microphone will emit only its own impulse response. Electrical spark discharges, pistol shots, and pressure-step methods all approximate the Dirac distribution, but due to their limitations one must still deconvolve the impulse responses of the excitation signal and that of the microphone itself. Since every known method for performing such deconvolution has further pitfalls of its own, a novel time-domain method of deconvolution is introduced.
Convention Paper 7065 (Purchase now)
P12-8 Toward Multimodal Interfaces for Intrusion Detection—Miguel Garcia-Ruiz, University of Colima - Colima, Mexico; Miguel Vargas Martin, Bill Kapralos, University of Ontario Institute of Technology - Oshawa, Ontario, Canada
Network intrusion detection has generally been dealt with using sophisticated software and statistical analysis tools. However, occasionally network intrusion detection must be performed manually by administrators, either by detecting the intruders in real-time or by revising network logs, making this a tedious and time consuming labor. To support this, intrusion detection analysis has been carried out using visual, auditory or tactile sensory information in computer interfaces. However, little is known about how to best integrate the sensory channels for analyzing intrusion detection. We propose a multimodal human-computer interface to analyze malicious attacks during forensic examination of network logs. We describe a sonification prototype that generates different sounds according to a number of well-known network attacks.
Convention Paper 7066 (Purchase now)
P12-9 Steganographic Approach to Copyright Protection of Audio—Suthikshn Kumar, PES Institute of Technology - Bangalore, India
Steganography is the technique of hiding data in images and music. It is one of the powerful mechanisms by which useful copyright information is hidden in the audio. In this paper we propose the use of steganography and public key cryptography to store the copyright information and authenticate the original audio. A tool called Steger is being developed that automatically determines the original copyright holders of the audio content. This tool is useful in Digital Rights Management (DRM) enabling end-user systems such as PDAs, mobile phones, PCs, handheld devices, consumer electronics, etc.
Convention Paper 7067 (Purchase now)