Jan Rennies

Listening effort and speech intelligibility in listening situations affected by noise and reverberation.

This study compared the combined effect of noise and reverberation on listening effort and speech intelligibility to predictions of the speech transmission index (STI). Listening effort was measured in normal-hearing subjects using a scaling procedure. Speech intelligibility scores were measured in the same subjects and conditions: (a) Speech-shaped noise as the only interfering factor, (b) + (c) fixed signal-to-noise ratios (SNRs) of 0 or 7 dB and reverberation as detrimental factors, and (d) reverberation as the only detrimental factor. In each condition, SNR and reverberation were combined to produce STI values of 0.17, 0.30, 0.43, 0.57, and 0.70, respectively. Listening effort always decreased with increasing STI, thus enabling a rough prediction, but a significant bias was observed indicating that listening effort was lower in reverberation only than in noise only at the same STI for one type of impulse responses. Accordingly, speech intelligibility increased with increasing STI and was significantly better in reverberation only than in noise only at the same STI. Further analyses showed that the broadband reverberation time is not always a good estimate of speech degradation in reverberation and that different speech materials may differ in their robustness toward detrimental effects of reverberation.

Comparison of loudness models for time-varying sounds

The loudness of a sound depends, among other parameters, on its temporal shape. Different loudness models were proposed to account for temporal aspects in loudness perception. To investigate different dynamic concepts for modeling loudness, predictions were made with the two current loudness models of Glasberg and Moore [J. Acoust. Soc. Am. 50, 331–341 (2002)] and Chalupper and Fastl [Acta Acustica united with Acustica 88, 378–386 (2002)] for a set of time-varying sounds. The predicted effects of duration, repetition rate, amplitude-modulation, temporal asymmetry, frequency modulation and the systematic variation of spectro-temporal structure on loudness were compared to data from the literature. Both models predicted the general trends of the data for single, repeated and asymmetric sound bursts and amplitude-modulated sounds. The model of Chalupper and Fastl seems to agree slightly better with loudness data for sounds with strong spectral variations over time, since it includes a dynamic stage which allows spectral loudness summation also for non-synchronous frequency components.

Effects of spatial and temporal integration of a single early reflection on speech intelligibilitya)

In order to study the interaction between the intelligibility advantage in rooms due to the presence of early reflections and due to binaural unmasking, a series of speech reception threshold experiments was performed employing a single reflection of the frontal target speech source as a function of its delay ranging from 0 to 200 ms. The direction of the reflection and the spatial characteristic of the interfering noise (diotic, diffuse, or laterally localized) were varied in the experiments. For the frontal reflection, full temporal integration was observed for all three noise types up to a delay of at least 25 ms followed by gradual intelligibility decay at longer delays. At 200 ms delay the reflection introduced additional intelligibility deterioration. For short delays, intelligibility was not reduced when the reflection was spatially separated from the direct sound in the diffuse and lateral noise conditions. A release from the deterioration effect at 200 ms delay was found for all spatially separated reflections. The suppression of a detrimental reflection was symmetrical in diffuse noise, but azimuth-dependent in lateral noise. This indicates an interaction of spatial and temporal processing of speech reflections which challenges existing binaural speech intelligibility models.

Suppression and comodulation masking release in normal-hearing and hearing-impaired listeners

The detectability of a sinusoidal signal embedded in a masker at the signal frequency can be improved by simultaneously presenting additional maskers in off-frequency regions if the additional maskers and the on-frequency masker component have the same temporal envelope. This effect is commonly referred to as comodulation masking release (CMR). Recently, it was hypothesized that peripheral nonlinear processes such as suppression may play a role in CMR over several octaves when the level of the off-frequency masker component is higher than the level of the on-frequency masker component. The aim of the present study was to test this hypothesis by measuring suppression and CMR within the same subjects for various frequency-level combinations of the off-frequency masker component. Experimental data for normal-hearing listeners show a large overlap between the existence regions for suppression and CMR. Hearing-impaired subjects with a sensorineural hearing loss show, on average, negligible suppression and CMR. The data support the hypothesis that part of the CMR in experiments with large spectral distances and large level differences between the masker components is due to the nonlinear processing at the level of the cochlea.

A study on speech quality and speech intelligibility measures for quality assessment of single-channel dereverberation algorithms

This paper reports on the evaluation of several objective quality measures for predicting the quality of the dereverberated speech signals. The correlations between subjective quality assessment for single-channel dereverberation techniques and objective speech quality as well as speech intelligibility measures are analyzed and discussed. Six different single-channel dereverberation algorithms were included in the evaluation to account for different types of distortions. The subjective quality was assessed along the four attributes reverberant, colored, distorted and overall quality following the recommendations of ITU-T P.835. The objective measures included system-based, i.e. channel-based, as well as signal-based measures.

Spectro-temporal weighting of loudness.

Real-world sounds like speech or traffic noise typically exhibit spectro-temporal variability because the energy in different spectral regions evolves differently as a sound unfolds in time. However, it is currently not well understood how the energy in different spectral and temporal portions contributes to loudness. This study investigated how listeners weight different temporal and spectral components of a sound when judging its overall loudness. Spectral weights were measured for the combination of three loudness-matched narrowband noises with different center frequencies. To measure temporal weights, 1,020-ms stimuli were presented, which randomly changed in level every 100 ms. Temporal weights were measured for each narrowband noise separately, and for a broadband noise containing the combination of the three noise bands. Finally, spectro-temporal weights were measured with stimuli where the level of the three narrowband noises randomly and independently changed every 100 ms. The data consistently showed that (i) the first 300 ms of the sounds had a greater influence on overall loudness perception than later temporal portions (primacy effect), and (ii) the lowest noise band contributed significantly more to overall loudness than the higher bands. The temporal weights did not differ between the three frequency bands. Notably, the spectral weights and temporal weights estimated from the conditions with only spectral or only temporal variability were very similar to the corresponding weights estimated in the spectro-temporal condition. The results indicate that the temporal and the spectral weighting of the loudness of a time-varying sound are independent processes. The spectral weights remain constant across time, and the temporal weights do not change across frequency. The results are discussed in the context of current loudness models.

Modeling Temporal Effects of Spectral Loudness Summation

Recent studies have shown that spectral loudness summation depends on duration. Modifications of a current loudness model were investigated with respect to their ability to predict this effect. The original version of the model could not simulate the duration dependence of spectral loudness summation. To reconcile the model with the loudness data, three different mechanisms accentuating temporal onsets of sounds were tested: (i) adaptive compression, (ii) adaptive auditory filters and (iii) bandwidth-dependent integration. A comparison between simulations and experimental data indicated that, in principle, all mechanisms lead to increased spectral loudness summation for short noise bursts, but bandwidth-dependent integration may be the most realistic approach. Such a modified model also predicts the spectral loudness summation of repeated noise bursts as a function of repetition rate.

Temporal weighting in loudness of broadband and narrowband signals

Temporal weights used by listeners when judging the overall loudness of a stimulus were measured for a 1-s-long noise centered around 2 kHz, whose level was randomly perturbed every 100 ms. The bandwidth was either 6400 Hz (broadband condition) or 400 Hz (narrowband condition). The first 100 ms contributed significantly more than later segments to overall loudness perception in the broadband condition. The effect was significantly reduced in the narrowband condition which is in line with the hypothesis that a greater spectral loudness summation at stimulus onset might be the mechanism behind the onset accentuation.

Speech-in-noise enhancement using amplification and dynamic range compression controlled by the speech intelligibility index

In many speech communication applications, such as public address systems, speech is degraded by additive noise, leading to reduced speech intelligibility. In this paper a pre-processing algorithm is proposed that is capable of increasing speech intelligibility under an equal-power constraint. The proposed AdaptDRC algorithm comprises two time- and frequency-dependent stages, i.e., an amplification stage and a dynamic range compression stage that are both dependent on the Speech Intelligibility Index (SII). Experiments using two objective measures, namely, the extended SII and the short-time objective intelligibility measure (STOI), and a formal listening test were conducted to compare the AdaptDRC algorithm with a modified version of a recently proposed algorithm in three different noise conditions (stationary car noise and speech-shaped noise and non-stationary cafeteria noise). While the objective measures indicate a similar performance for both algorithms, results from the formal listening test indicate that for the two stationary noises both algorithms lead to statistically significant improvements in speech intelligibility and for the non-stationary cafeteria noise only the proposed AdaptDRC algorithm leads to statistically significant improvements. A comparison of both objective measures and results from the listening test shows high correlations, although, in general, the performance of both algorithms is overestimated.

Hands-free telecommunication for elderly persons suffering from hearing deficiencies

Speech communication is the most natural form of human interaction. Communication by means of telephones, mobile phones or video-conference systems is common nowadays especially amongst younger persons. In the past years, also a growing amount of elderly people has started to extensively use communication systems since more and more people live apart from their relatives, friends or acquaintances. However, especially elderly people suffer from hearing loss, which often prevents them from using acoustic communication devices. While approximately every second European adult of age 65+ has a hearing loss that requires treatment, only the minority actually wears hearing aids for different reasons. To tackle this problem, this contribution deals with a personalized and adaptable communication system that enhances the acoustic signal and incorporates the individual hearing loss of a hearing-impaired person. By this, the typical elderly user is enabled to take part in natural communication again.