Journal of the Audio Engineering Society

2005 October - Volume 53 Number 10


Long Interpolation of Audio Signals Using Linear Prediction in Sinusoidal Modeling

Authors: Lagrange, Mathieu; Marchand, Sylvain; Rault, Jean-bernard

Within the context of sinusoidal modeling, a new method for the interpolation of sinusoidal components is proposed. It is shown that autoregressive modeling of the amplitude and frequency parameters of these components allows us to interpolate missing audio data realistically, especially in the case of musical modulations such as vibrato or tremolo. The problem of phase discontinuity at the gap boundaries is also addressed. Finally, an original algorithm for the interpolation of a missing region of a whole set of sinusoids is presented. Objective and subjective tests show that the quality is improved significantly compared to common sinusoidal and temporal interpolation techniques of missing audio data.

A model for predicting the audibility of time-varying signals in background sounds is described. The model requires the calculation of time-varying excitation patterns for the signal and background, using the methods described elsewhere. A quantity called instantaneous partial loudness (IPL) is calculated from the excitation patterns. The estimates of IPL, which are updated every 1 ms, are used to calculate the short-term partial loudness (STPL) using a form of running average similar to an automatic gain control system. It is assumed that the audibility of the signal is monotonically related to the average value of the STPL over the duration of the signal. In experiment 1 thresholds were measured for detecting a 1-kHz sinusoid in four different samples each of white and pink “frozen” noise. The results were used to determine the average value of the STPL required for threshold. In experiment 2 the model was evaluated by measuring detection thresholds for nine signal types in six backgrounds (54 combinations), using a two-alternative forced-choice task. The backgrounds were chosen to be relatively steady (such as traffic noise). The correlation between the measured masked thresholds and those predicted by the model was 0.94. The root-meansquare difference between the thresholds obtained and those predicted was 3 dB. In experiment 3 psychometric functions were measured for the detection of five signals in five backgrounds (five pairs), using a two-alternative forced-choice task. Experiment 4 used the same signals and backgrounds, but psychometric functions were measured using a singleinterval yes–no task. The results of experiments 3 and 4 were used to construct functions relating signal detectability d to the average value of the STPL.

Directional Resolution of Head-Related Transfer Functions Required in Binaural Synthesis

Authors: Minnaar, Pauli; Plogsties, Jan; Christensen, Flemming

In binaural synthesis a virtual sound source is implemented by convolving an anechoic signal with a pair of head-related transfer functions (HRTFs). In order to represent all possible directions of the sound source with respect to the listener a discrete number of HRTFs are measured and interpolations are made in between. A listening experiment was done to estimate the lowest directional resolution with which HRTFs have to be measured to ensure that interpolations between them do not introduce audible errors. In order to make this study the HRTFs of an artificial head were measured with a directional resolution of 2°. The measurements were used to create HRTF data sets with low resolution from which interpolations were made in the horizontal, frontal, and median planes. Measured and interpolated HRTFs were compared in a three-alternative forced-choice listening experiment for both stationary and moving sound sources. A criterion was found that predicts the experimental results. This criterion was used to estimate the directional resolution required in binaural synthesis for all directions on the sphere around the head.

A new acoustic filter that uses four polymer films with half-wavelength spacing was designed and used for detailed measurements of the audio signal components from amplitude-modulated high-intensity ultrasonic waves. The filter provided strong attenuation of the carrier frequency, which is a major cause of microphone nonlinearity. The measurements clearly showed parametric array effects from 300 to 8000 Hz and modulated radiation pressure above 30 Hz.

Program Loudness Revisited

Authors: Staff, AES

[Feature Article] In June 2004 we reported on a number of attempts to tackle the problem of measuring and controlling the loudness of broadcast programs. Here we provide a short update on the topic, as represented in papers from the AES 118th Convention in Barcelona.

[Feature Article] The world of film sound postproduction is changing fast. No longer is the process sequential and clear-cut (some might say it never was, of course). Now the sound track of a film can be seen to emerge from the melting pot of the production process, being created almost concurrently with the picture edit. Whereas in former days a finished (“locked off”) picture cut might be handed over to the sound department to work on, it can now be changed right up until the last moment because picture editing is undertaken digitally using a computer. This new-found freedom is fully exploited by visual-effects departments and picture editors, leaving the sound editor to follow along as best she can and make a convincing sound track from a picture cut that changes every day. Two papers from the 118th Convention highlight aspects of this changing world, showing how new technology has significantly changed the task facing sound editors, composers, and orchestrators.

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Program Loudness Revisited

The Evolving World of Film Sound

New Officers 2005/2006

120th Convention, Paris, Call for Papers


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