Comparative evaluation of predicted and measured performance of a 68-cubic meter truncated reverberant noise chamber

The performance of a medium size, truncated reverberation chamber is evaluated in detail. Chamber performance parameters are predicted, using classical acoustic theory, and are compared to results from actual chamber measurements. Discrepancies are discussed in relation to several available empirical corrections developed by other researchers. Of more practical interest is the confirmation of a recent theory stating that the present guide for the ratio of specimen volume to test chamber volume, approximately 10 percent, is overly conservative, and can be increased by a factor of at least 2 and possibly 3. Results and theoretical justification of these findings are presented

Effect of Vertical Microphone Layer Spacing for a 3D Microphone Array

Subjective listening tests were conducted to investigate how the spacing between main (lower) and height (upper) microphone layers in a 3D main microphone array affects perceived spatial impression and overall preference. Four different layer spacings of 0m, 0.5m, 1m, and 1.5m were compared for the sound sources of trumpet, acoustic guitar, percussion quartet, and string quartet using a nine-channel loudspeaker setup. It was generally found that there was no significant difference between any of the spaced layer configurations, whereas the 0m layer had slightly higher ratings than the more spaced layers in both spatial impression and preference. Acoustical properties of the original microphone channel signals as well as those of the reproduced signals, which were binaurally recorded, were analyzed in order to find possible physical causes for the perceived results. It is suggested that the perceived results were mainly associated with vertical interchannel crosstalk in the signals of each height layer and the magnitude and pattern of spectral change at the listener’s ear caused by each layer

Improving speech intelligibility in hearing aids. Part I: Signal processing algorithms

[EN] The improvement of speech intelligibility in hearing aids is a traditional problem that still remains open and unsolved. Modern devices may include signal processing algorithms to improve intelligibility: automatic gain control, automatic environmental classification or speech enhancement. However, the design of such algorithms is strongly restricted by some engineering constraints caused by the reduced dimensions of hearing aid devices. In this paper, we discuss the application of state-of-theart signal processing algorithms to improve speech intelligibility in digital hearing aids, with particular emphasis on speech enhancement algorithms. Different alternatives for both monaural and binaural speech enhancement have been considered, arguing whether they are suitable to be implemented in a commercial hearing aid or not.This work has been funded by the Spanish Ministry of Science and Innovation, under project TEC2012-38142-C04-02.Ayllón, D.; Gil Pita, R.; Rosa Zurera, M.; Padilla, L.; Piñero Sipán, MG.; Diego Antón, MD.; Ferrer Contreras, M.... (2014). Improving speech intelligibility in hearing aids. Part I: Signal processing algorithms. Waves. 6:61-71. http://hdl.handle.net/10251/57901S6171

Evaluation of directional microphone drift in digital hearing aids

The occurrence of directional microphone drift following hearing aid use has been infrequently examined. This study uses the front-to-side ratio to evaluate changes in directional microphone output from new behind-the-ear hearing aids and following approximately three months of hearing aid use. Results indicate no overall significant differences in the front-to-side ratio between initial and follow-up measurements