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P25 - Instrumentation and MeasurementTuesday, May 23, 09:00 — 11:40
Chair: Michel Keyhl, OPTICOM - Erangen, Germany
P25-1 Automatic Recognition of Urban Sound Sources—Boris DeFreville, LASA - Paris, France, University of Cergy-Pontoise, Cergy-Pontoise, France; Pierre Roy, Sony CSL Paris - Paris, France; Christophe Rosin, LASA - Paris, France; François Pachet, SONY CSL Paris - Paris, France
The goal of the FDAI project is to create a general system that computes an efficient representation of the acoustic environment. More precisely, FDAI has to compute a noise disturbance indicator based on the identification of six categories of sound sources. This paper describes experiments carried out to identify acoustic features and recognition models that were implemented in FDAI. This framework is based on EDS–Extractor Discovery System–an innovative acoustic feature extraction system for sound feature extraction. The design and development of FDAI raised two critical issues. Completeness: it is very difficult to design descriptors that identify every sound source in urban environments; and Consistency: some sound sources are not acoustically consistent. We solved the first issue with a conditional evaluation of a family of acoustic descriptors, rather than the evaluation of a single general-purpose extractor. Indeed, a first hierarchical separation between vehicles (moped, bus, motorcycle and car) and non-vehicles (bird and voice) significantly raised the accuracy of identification of the buses. The second issue turned out to be more complex and is still under study. We give here preliminary results.
Presentation is scheduled to begin at 09:00
Convention Paper 6827 (Purchase now)
P25-2 A New Integrated System for Laboratory Speech/Voice Examination—Costas Pastiadis, Aristotle University of Thessaloniki - Thessaloniki, Greece; Georgia Psyllidou, Paris Telecom - Paris, France; George Papanikolaou, Aristotle University of Thessaloniki - Thessaloniki, Greece
The paper presents a new computer-based system for the examination and analysis of speech/voice functionality in laboratory environments. Although the system is mainly designed for clinical applications, it employs features that afford its generalized use as a speech/voice acquisition, analysis, and evaluation tool. The system offers an integrated and interactive modular structure for the conduction of various speech/voice examination procedures, and provides necessary data management capabilities for further exploitation in diagnostic expert systems and knowledge-based speech/voice applications.
[Associated Poster Presentation in Session P30, Tuesday, May 23, at 14:00]
Presentation is scheduled to begin at 09:20
Convention Paper 6828 (Purchase now)
P25-3 Directivity Measurements on a Highly Directive Hearing Aid: The Hearing Glasses—Marinus M. Boone, Technical University of Delft - Delft, The Netherlands
A highly directional hearing aid has been developed with the aim to give a much higher speech intelligibility than with conventional hearing aids. The high directivity is obtained by mounting four microphones in each temple of a pair of glasses and performing optimized beam forming. This leads to an averaged directivity index of 9 dB under free field conditions, without head disturbance. In a recent research program the directivity of this device has been measured with different directivity settings under free field and diffuse field conditions, with and without head diffraction. Results are presented of this research, where a comparison will also be made with the directivity of a conventional hearing aid. Also, the influence of the setting of the superdirective beamforming on the noise sensitivity is shown, indicating that for practical use the directivity should be limited.
[Associated Poster Presentation in Session P30, Tuesday, May 23, at 14:00]
Presentation is scheduled to begin at 09:40
Convention Paper 6829 (Purchase now)
P25-4 Accurate Nonlinear Models of Valve Amplifiers Including Output Transformers—Pierre Touzelet, Technical Director - Vélizy, France
Available commercial network analysis programs are now powerful enough to look at sophisticated models of complete valve amplifiers including nonlinear components such as valves and output transformers. Objectives of such accurate nonlinear models are evident. They allow for the evaluation, with a high degree of realism, of global amplifier performances and their distortion, reducing, as a result, major risks at the development stage of any amplifier project. It is the intention of this paper to show how such sophisticated models can be developed and which kind of results and information can be extracted from them, by applying these sophisticated modelizations on a real amplifier, as an illustrative example.
Presentation is scheduled to begin at 10:00
Convention Paper 6830 (Purchase now)
P25-5 The Self-Compensated Audio Transformers for Tube and Solid State Single-Ended Amplifiers—Aristide Polisois, A2B Electronic - La Houssaye en Brie, France; Giovanni Mariani, GRAAF srl - Modena, Italy
The self-compensated output transformer presented at the AES Convention held in Barcelona in May 2005 (Convention Paper 6346), intended for single-ended audio amplifiers, is based on the principle that an auxiliary winding (named tertiary), crossed by the same current as the primary winding, can oppose a magnetic flux that reduces the overall flux, produced by the direct current, in the core, to almost zero. However, at the same time, this antagonist winding also opposes the induced alternating current. A capacitor is therefore connected to its terminals, short-circuiting the alternating current. Under these circumstances, the alternating potential difference is close to zero and the primary is no less affected. But the above short-circuit has a drawback: it reduces the inductance of the primary, considerably. Novel solutions have been found to remove this obstacle to a satisfactory performance of the self-compensated output transformer.
[Associated Poster Presentation in Session P30, Tuesday, May 23, at 14:00]
Presentation is scheduled to begin at 10:20
Convention Paper 6831 (Purchase now)
P25-6 Some Neglected Audio Distortion Mechanisms—Richard Black, Richard Black Associates - London, UK
In addition to the familiar harmonic and intermodulation distortions, there exist various other mechanisms by which electronic equipment can degrade sound quality. Some of these are closely related to the familiar types, others are the result of direct acoustical interaction of the equipment, while yet others rely on the existence of two (or more) unrelated distortions in a system to produce an audible result. This paper examines some of these distortion mechanisms.
Presentation is scheduled to begin at 10:40
Convention Paper 6832 (Purchase now)
P25-7 Comparison of Four Subwoofer Measurement Techniques—Manuel Melon, Christophe Langrenne, CNAM, Laboratoire d’Acoustique - Paris Cedex, France; David Rousseau, Bruno Roux, BC Acoustique - Alfortville, France; Philippe Herzog, Laboratoire de Mécanique et D’Acoustique - Marsielle, France
Acoustic measurements at very low frequency are difficult to perform. Then, interpretation of the results is tricky. In this paper four subwoofer measurement techniques are compared in terms of frequency response and directivity. The methods used are the following ones: anechoic room, pseudo free-field, isobaric room, and semi-anechoic room. Three subwoofers are tested: two closed-box systems and an active/passive system. For the semi-anechoic technique, double layer pressure measurements on a half-sphere surrounding the source are performed. Then, using spherical harmonic decomposition, outgoing and ingoing pressure fields are separated to recover free field conditions (i.e., removal of reflections on walls below room cut-off frequency). Discrepancies between results are discussed and explained when possible.
Presentation is scheduled to begin at 11:00
Convention Paper 6833 (Purchase now)
P25-8 Room Impulse Response Measurement Using a Moving Microphone—Thibaut Ajdler, Luciano Sbaiz, Ecole Polytechnique Fédérale de Lausanne (EPFL) - Lausanne, Switzerland; Martin Vetterli, Ecole Polytechnique Fédérale de Lausanne (EPFL) - Lausanne, Switzerland, and University of California at Berkeley, Berkeley, CA, USA
In this paper we present a technique to record a large set of room impulse responses using a microphone moving along a trajectory. The technique processes the signal recorded by the microphone to reconstruct the signals that would have been recorded at all the possible spatial positions along the array. The speed of movement of the microphone is shown to be the key factor for the reconstruction. This fast method of recording spatial impulse responses can also be applied for the recording of head-related transfer functions.
[Associated Poster Presentation in Session P30, Tuesday, May 23, at 14:00]
Presentation is scheduled to begin at 11:20
Convention Paper 6834 (Purchase now)
