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Journal of the Audio Engineering Society

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2002 May - Volume 50 Number 5

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Papers

Audio for Games

Authors: Wilde, Martin

Not available.

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Managing Change: The Challenge of Rights Management in the New Millennium

Authors: Hill, Keith

This article will identify and discuss rights management issues that have to be solved in the digital domain to ensure the continued viability of music-related commerce. It discusses the principles of copyright ownership and how they may be upheld through the development and evolution of information management standards.

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Dipole Loudspeaker Response in Listening Rooms

Authors: Kates, James M.

A dipole loudspeaker radiates from both its front and its rear surfaces, with the radiation from the rear 180° out of phase with the radiation from the front. A conventional monopole loudspeaker is a pressure source, but a dipole loudspeaker is a velocity source. The dipole frequency response in the room thus depends on orientation as well as on room location. The directional pattern of a dipole, however, can provide better auditory localization than a monopole when the dipole is angled into the room. A simulation study is presented of the effects of room placement and orientation angle on the frequency response of a dipole loudspeaker in a room. The frequency response using a two-dimensional room model is computed, and a perceptual model is used to estimate the importance of the room response effects in coloring the reproduced sound. The simulation results indicate that the response of a dipole is more sensitive to changes in room position than that of a monopole, and that it is also sensitive to the angle of orientation within the room.

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An IIR Synthesis Method for Plucked-String Instruments with Embedded Portamento

Authors: Su, Alvin W. Y.; Chang, Wei-Chen; Wang, Rei-Wen

Portamento and vibrato are the two most used playing techniques for string instruments. For some specific instruments, such as the Gu-Chin (a Chinese traditional zither), the range of portamento can be close to an octave. The sound becomes unnatural in such cases if wavetable techniques are used. A new infinite-impulse-response (IIR) synthesis model with embedded portamento is proposed. This method can closely synthesize a specific plucked-string instrument and also perform portamento and vibrato efficiently. The model parameters for synthesizing a specific instrument are obtained by using a neural network training algorithm. A variable-length delay line is embedded such that the special effects can be obtained by changing the system parameters in resynthesis. The proposed method has been successfully tested over many different kinds of plucked-string instruments. The computational complexity of synthesis is close to that of an ordinary IIR filter and depends on its order. The complexity of the proposed model is higher if more accuracy is desired. Like DWF-based approaches, memory is required to store a small segment of excitation waveform.

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Perception of Reverberation Time in Small Listening Rooms

Authors: Niaounakis, T. I.; Davies, William J.

The perceptibility of small changes in reverberation time (RT) when music is reproduced within small studio control rooms was judged by a number of subjects. The aim of the study was to determine the difference limen (DL) for midfrequency RTs shorter than 0.6 s, which are usually encountered in such rooms. The work used a real control room and consisted of two experiments. In the first experiment the DL for RT was measured by changing the amount of absorption in the room. Here the measurement stimulus (music) was reproduced to the subjects by one loudspeaker located in the center in front of the mixing console. In the second experiment the measurement stimuli were recorded in the control room using a dummy head and later replayed to the subjects by headphones. Differences between the two measurements were compared and the overall DL was found to be 0.042 6 0.015 s.

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A Model of Loudness Applicable to Time-Varying Sounds

Authors: Glasberg, Brian R.; Moore, Brian C. J.

Previously we described a model for calculating the loudness of steady sounds from their spectrum. Here a new version of the model is presented, which uses a waveform as its input. The stages of the model are as follows. (a) A finite impulse response filter representing transfer through the outer and middle ear. (b) Calculation of the short-term spectrum using the fast Fourier transform (FFT). To give adequate spectral resolution at low frequencies, combined with adequate temporal resolution at high frequencies, six FFTs are calculated in parallel, using longer signal segments for low frequencies and shorter segments for higher frequencies. (c) Calculation of an excitation pattern from the physical spectrum. (d) Transformation of the excitation pattern to a specific loudness pattern. (e) Determination of the area under the specific loudness pattern. This gives a value for the "instantaneous" loudness. The short-term perceived loudness is calculated from the instantaneous loudness using an averaging mechanism similar to an automatic gain control system, with attack and release times. Finally the overall loudness impression is calculated from the short-term loudness using a similar averaging mechanism, but with longer attack and release times. The new model gives very similar predictions to our earlier model for steady sounds. In addition, it can predict the loudness of brief sounds as a function of duration and the overall loudness of sounds that are amplitude modulated at various rates.

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