Authors:Giannoulis, Dimitrios; Massberg, Michael; Reiss, Joshua D.
Affiliation:Queen Mary University of London, London, UK
Unlike common digital signal processing algorithms, dynamic range compressors are nonlinear systems with memory. Furthermore, there is no single established design approach. Thus, they allow for a large number of implementations, each with very different behavior. Following an analysis of the various design choices, the authors recommend feed-forward algorithms because they are stable and predictable. The detector is placed in the log domain after the gain computer since this generates a smooth envelope, no attack lag, and a variable knee width. Objective measures of compressor performance are also used and discussed, and source code and audio examples are available on line.
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Authors:Parseihian, Gaëtan; Katz, Brian F. G.
Affiliation:LIMSI-CNRS, Orsay, France
Sonification, a form of auditory display, is a means of mapping arbitrary information such as the distance to an obstacle or changes in temperature into sound. For the visually impaired, sonification can make an important contribution to increasing autonomy. In contrast to earcons (the audio analog of icons), which map a unique sound to a particular meaning, morphocons are short audio units that are used to construct a sonic grammar based on temporal-frequency patterns, rather than fixed sound samples. For example, a rhythmic repetition can be used to modify or add meaning to any base sound sample. Results indicate that both blind and sighted subjects were able to perceive temporal variations of acoustic parameters as an abstract form, independent of the base sound sample, allowing the extraction of consistent category information from a range of different customizable sounds.
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Authors:Cerdá, Salvador; Giménez, Alicia; Cibrián, Rosa M.
Affiliation:Universitat Politécnica de Valéncia, Valencia, Spain; Universitat de Valéncia, Valencia, Spain
Producing a formal means for ranking the quality of concert halls requires that the subjective assessment by experts and listeners be matched to acoustic parameters. Using previous results from Ando and Beranek, the authors propose a statistical scheme to obtain a function that fits the objective experimental data from 17 performance halls, which vary in uses, sizes, building types, and acoustics. Of the many possible acoustic parameters available, it appears that most of them can be reduced to LEV (listener envelopment), RTmid (mid frequency reverberation time), and LFCE4 (early lateral energy fraction). The obtained model based on these parameters, allows for designing and improving performance spaces.
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Authors:Samaali, Imen; Mahé, Gaël; Turki, Monia
Affiliation:LIPADE, University Paris Descartes, Sorbonne Paris Cité, France; U2S, National Engineering School of Tunis, University Tunis, El Manar, Tunisia
Low bit-rate coding/decoding with standard AAC and MP3 coders smoothes attacks in transient signals and increases the pre-echo. A system is proposed for reducing pre-echo artifacts by embedding temporal envelope correction as watermarking data, which eliminates the need for an auxiliary communications channel. The restored signal is reconstructed at the receiver from two components: the decoded signal generated by the core decoder, and the temporal envelope of the original signal, which is represented by a few parameters that are transmitted as embedded data in the audio signal.
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Authors:Huckvale, Mark; Hilkhuysen, Gaston
Affiliation:University College London, London, UK
Speech communication systems need to be evaluated across a wide range of signal qualities. However, when signal quality is high, evaluations focus on listener acceptance rather than on task performance because conventional intelligibility tests reach ceiling values. This research considers whether measures of cognitive load could be used to measure the effect of signal quality on communication performance even when intelligibility is high. It is shown that differences in signal quality can affect the ability of listeners to detect transcription errors in an audio proof-reading task. The research also shows that a noise reduction system, which elsewhere has been said to improve listener acceptance, gives no improvement in terms of cognitive load on this task.
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Affiliation:KLIPPEL GmbH, Dresden, Germany
An accurate means for measuring the magnetic flux density of moving coil loudspeakers is critically important. The paper presents a new measurement technique by scanning flux density on a cylindrical surface within and outside the magnetic gap using a Hall sensor combined with robotics to move the sensor. Scanning the magnet field on the circumference of the gap seems to be the only way for detecting axial asymmetries and other magnetic field irregularities caused by material inhomogeneity and manufacturing failures, such as partial magnetization and incorrect adjustment of the parts. Variation of magnetic field in the gap reveals an asymmetrical driving force that causes rocking modes and voice coil rubbing.
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As digital signal processing becomes more powerful, there is increasing scope for handling audio signals in the frequency domain—dividing the signal into narrow frequency bands and performing operations on those bands independently. This is not a new concept, having been used in applications such as low bitrate coding for many years, but the amount of work in the field has become so great that the AES organized a conference recently in Finland, chaired by Ville Pulkki, devoted entirely to the topic. This article attempts to condense some of the contributions from that conference, taking what is essentially a heavily mathematical topic and presenting it in a more digestible form.
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