Jasper van Dorp Schuitman

Development of a clarity parameter using a time-varying loudness model.

The perceived sound clarity is often estimated with the clarity index, which is calculated on the basis of physical acoustic measures that can correlate weakly to the way humans perceive sound for certain test conditions. Therefore, this study proposes a clarity parameter based on a binaural room impulse response processed with a time-varying loudness model. The proposed parameter is validated by calculating the correlation coefficient with subject responses collected from previous listening experiments. Results show that the parameter outperforms the clarity index in most of the tested conditions, but its performance is less robust than parameter for clarity (PCLA).

Comparison of psychoacoustic-based reverberance parameters

This study compared psychoacoustic reverberance parameters to each other, as well as to reverberation time (RT) and early decay time (EDT) under various acoustic conditions. The psychoacoustic parameters were loudness-based RT (TN), loudness-based EDT [EDTN; Lee, Cabrera, and Martens, J. Acoust. Soc. Am. 131, 1194-1205 (2012a)], and parameter for reverberance [PREV; van Dorp Schuitman, de Vries, and Lindau., J. Acoust. Soc. Am. 133, 1572-1585 (2013)]. For the comparisons, a wide range of sound pressure levels (SPLs) from 20u2009dB to 100u2009dB and RTs from 0.5u2009s to 5.0u2009s were evaluated, and two sets of subjective data from the previous studies were used for the cross-validation and comparison. Results of the comparisons show that the psychoacoustic reverberance parameters provided better matches to reverberance than RT and EDT; however, the performance of these psychoacoustic reverberance parameters varied with the SPL range, the type of audio sample, and the reverberation conditions. This study reveals that PREV is the most relevant for estimating a relative change in reverberance between samples when the SPL range is small, while EDTN is useful in estimating the absolute reverberance. This study also suggests the use of PREV and EDTN for speech and music samples, respectively.

Perceptual limits for detecting interaural-cue manipulations measured in reverberant settings

In this study, binaural room impulse responses (BRIRs) were manipulated to determine the just noticeable differences in the interaural time delay (ITD), interaural level difference (ILD), and interaural cross-correlation (ICC) in reverberant settings. The BRIR were split in two sections, the first 75–150 ms of the BRIR was found to be direction dependent, and for this first part either an extra ITD or ILD was applied. These manipulations were expected to change the perceived direction of the sound source. Changes in the ICC were applied to the remaining part of the BRIR, which was expected to change the overall spatial impression, but not the perceived location. Each of these three differently manipulated BRIRs was convolved with an anechoic musical instrument, and the just noticeable change in ITD, ILD, or ICC was determined in a listening experiment. Due to the convolution with a temporally varying musical instrument stimulus, a complex spectrotemporal pattern of binaural cues is created. An analysis of...

Modelling of the cochlea response as a versatile tool for acoustic signal processing

The inner ear or cochlea processes the acoustic signals that enter the oval window into a specific time‐frequency pattern. Many acoustic signal processing methods are based on this behaviour. A fundamental method is to calculate this time‐frequency response by solving the differential equation of the movement of the basilar membrane, followed by a visualisation of the excitation patterns in a time‐frequency plot. For that purpose Continuity Preserving Signal Processing (CPSP) is a promising method. In the presentation an overview will be given of a project that is carried out by TUD (University of Technology Delft) together with RUG (University of Groningen) being sponsored by STW (Dutch Technology Foundation). The project divides into four sub‐projects which are closely related: Automatic Keyword Spotting, Machine Analysis and Diagnostics, Speech Intelligibility Enhancement for Hearing Aids and Quality Assessment of Room Acoustics. Results that have been obtained in the project will be summarised. Detailed results of the sub‐projects will be presented in separate presentations.

Determining acoustical parameters using cochlear modeling and auditory masking

The acoustical qualities of a concert hall or any other room are generally expressed using acoustical parameters determined from impulse responses. From microphone array measurements it turned out that these parameters can fluctuate severely over small distances, whereas the perceptual cues for which these parameters are supposed to be a measure remain constant. This means that a local parameter value has a very low predictive value for acoustic quality. In this research, cochlear modeling techniques and simulations of auditory masking effects have been applied to model human hearing. These techniques together model various stages in the auditory path, like the movement of the basilar membrane inside the cochlea and mechanisms inside the brains. It turns out that determining acoustical parameters using this representation leads to results which show much less spatial fluctuations, and are closer to human perception.

A new digital module for variable acoustics and wave field synthesis

In the 1980s, wave field synthesis (WFS) was proposed by Berkhout [J. Audio Eng. Soc., 36, pp. 977–995 (1988)] as a new and powerful concept for spatial reproduction of sound fields without sweet‐spot‐limitations. Based on this concept, with implementation of simplifications not deteriorating the perceptual quality, a system for variable acoustics was developed: acoustic control system (ACS). The system has been widely applied as a solution for multipurpose halls. Until now, ACS processors have been built using analog electronic technology. Recently, however, a digital module has been designed that not only enables us to generate the ACS reflection patterns, but also to perform true WFS, e.g., for direct sound reinforcement with preservation of natural spatial properties. In the paper, design and specifications of the module will be discussed. Also, the potentials of integrated application of WFS and variable acoustics will be explained.