Vehicle noise: comparison of loudness ratings in the field and the laboratory

OBJECTIVE

Distorted loudness perception is one of the main complaints of hearing aid users. Measuring loudness perception in the clinic as experienced in everyday listening situations is important for loudness-based hearing aid fitting. Little research has been done comparing loudness perception in the field and in the laboratory.

DESIGN

Participants rated the loudness in the field and in the laboratory of 36 driving actions. The field measurements were recorded with a 360° camera and a tetrahedral microphone. The recorded stimuli, which are openly accessible, were presented in three conditions in the laboratory: 360° video recordings with a head-mounted display, video recordings with a desktop monitor and audio-only.

STUDY SAMPLES

Thirteen normal-hearing participants and 18 hearing-impaired participants with hearing aids.

RESULTS

The driving actions were rated as louder in the laboratory than in the field for the condition with a desktop monitor and for the audio-only condition. The less realistic a laboratory condition was, the more likely it was for a participant to rate a driving action as louder. The field-laboratory loudness differences were bigger for louder sounds.

CONCLUSION

The results of this experiment indicate the importance of increasing realism and immersion when measuring loudness in the clinic.

The future of hearing aid technology : Can technology turn us into superheroes?

BACKGROUND

Hearing aid technology has proven to be successful in the rehabilitation of hearing loss, but its performance is still limited in difficult everyday conditions characterized by noise and reverberation.

OBJECTIVE

Introduction to the current state of hearing aid technology and presentation of the current state of research and future developments.

METHODS

The current literature was analyzed and several specific new developments are presented.

RESULTS

Both objective and subjective data from empirical studies show the limitations of the current technology. Examples of current research show the potential of machine learning-based algorithms and multimodal signal processing for improving speech processing and perception, of using virtual reality for improving hearing device fitting and of mobile health technology for improving hearing health services.

CONCLUSION

Hearing device technology will remain a key factor in the rehabilitation of hearing impairments. New technology, such as machine learning and multimodal signal processing, virtual reality and mobile health technology, will improve speech enhancement, individual fitting and communication training, thus providing better support for all hearing-impaired patients, including older patients with disabilities or declining cognitive skills.

Challenging Times for Cochlear Implant Users - Effect of Face Masks on Audiovisual Speech Understanding during the COVID-19 Pandemic

Unhindered auditory and visual signals are essential for a sufficient speech understanding of cochlear implant (CI) users. Face masks are an important hygiene measurement against the COVID-19 virus but disrupt these signals. This study determinates the extent and the mechanisms of speech intelligibility alteration in CI users caused by different face masks. The audiovisual German matrix sentence test was used to determine speech reception thresholds (SRT) in noise in different conditions (audiovisual, audio-only, speechreading and masked audiovisual using two different face masks). Thirty-seven CI users and ten normal-hearing listeners (NH) were included. CI users showed a reduction in speech reception threshold of 5.0 dB due to surgical mask and 6.5 dB due to FFP2 mask compared to the audiovisual condition without mask. The greater proportion of reduction in SRT by mask could be accounted for by the loss of the visual signal (up to 4.5 dB). The effect of each mask was significantly larger in CI users who exclusively hear with their CI (surgical: 7.8 dB, p = 0.005 and FFP2: 8.7 dB, p = 0.01) compared to NH (surgical: 3.8 dB and FFP2: 5.1 dB). This study confirms that CI users who exclusively rely on their CI for hearing are particularly susceptible. Therefore, visual signals should be made accessible for communication whenever possible, especially when communicating with CI users.

Synchronization of ear-EEG and audio streams in a portable research hearing device

Recent advancements in neuroscientific research and miniaturized ear-electroencephalography (EEG) technologies have led to the idea of employing brain signals as additional input to hearing aid algorithms. The information acquired through EEG could potentially be used to control the audio signal processing of the hearing aid or to monitor communication-related physiological factors. In previous work, we implemented a research platform to develop methods that utilize EEG in combination with a hearing device. The setup combines currently available mobile EEG hardware and the so-called Portable Hearing Laboratory (PHL), which can fully replicate a complete hearing aid. Audio and EEG data are synchronized using the Lab Streaming Layer (LSL) framework. In this study, we evaluated the setup in three scenarios focusing particularly on the alignment of audio and EEG data. In Scenario I, we measured the latency between software event markers and actual audio playback of the PHL. In Scenario II, we measured the latency between an analog input signal and the sampled data stream of the EEG system. In Scenario III, we measured the latency in the whole setup as it would be used in a real EEG experiment. The results of Scenario I showed a jitter (standard deviation of trial latencies) of below 0.1 ms. The jitter in Scenarios II and III was around 3 ms in both cases. The results suggest that the increased jitter compared to Scenario I can be attributed to the EEG system. Overall, the findings show that the measurement setup can time-accurately present acoustic stimuli while generating LSL data streams over multiple hours of playback. Further, the setup can capture the audio and EEG LSL streams with sufficient temporal accuracy to extract event-related potentials from EEG signals. We conclude that our setup is suitable for studying closed-loop EEG & audio applications for future hearing aids.

How Face Masks Interfere With Speech Understanding of Normal-Hearing Individuals: Vision Makes the Difference

OBJECTIVE

To investigate the effects of wearing a simulated mask on speech perception of normal-hearing subjects.

STUDY DESIGN

Prospective cohort study.

SETTING

University hospital.

PATIENTS

Fifteen normal-hearing, native German speakers (8 female, 7 male).

INTERVENTION

Different experimental conditions with and without simulated face masks using the audiovisual version of the female German Matrix test (Oldenburger Satztest, OLSA).

MAIN OUTCOME MEASURES

Signal-to-noise ratio (SNR) at speech intelligibility of 80%.

RESULTS

The SNR at which 80% speech intelligibility was achieved deteriorated by a mean of 4.1 dB SNR when simulating a medical mask and by 5.1 dB SNR when simulating a cloth mask in comparison to the audiovisual condition without mask. Interestingly, the contribution of the visual component alone was 2.6 dB SNR and thus had a larger effect than the acoustic component in the medical mask condition.

CONCLUSIONS

As expected, speech understanding with face masks was significantly worse than under control conditions. Thus, the speaker's use of face masks leads to a significant deterioration of speech understanding by the normal-hearing listener. The data suggest that these effects may play a role in many everyday situations that typically involve noise.

Making sense of periodicity glimpses in a prediction-update-loop-A computational model of attentive voice tracking

Humans are able to follow a speaker even in challenging acoustic conditions. The perceptual mechanisms underlying this ability remain unclear. A computational model of attentive voice tracking, consisting of four computational blocks: (1) sparse periodicity-based auditory features (sPAF) extraction, (2) foreground-background segregation, (3) state estimation, and (4) top-down knowledge, is presented. The model connects the theories about auditory glimpses, foreground-background segregation, and Bayesian inference. It is implemented with the sPAF, sequential Monte Carlo sampling, and probabilistic voice models. The model is evaluated by comparing it with the human data obtained in the study by Woods and McDermott [Curr. Biol. 25(17), 2238-2246 (2015)], which measured the ability to track one of two competing voices with time-varying parameters [fundamental frequency (F0) and formants (F1,F2)]. Three model versions were tested, which differ in the type of information used for the segregation: version (a) uses the oracle F0, version (b) uses the estimated F0, and version (c) uses the spectral shape derived from the estimated F0 and oracle F1 and F2. Version (a) simulates the optimal human performance in conditions with the largest separation between the voices, version (b) simulates the conditions in which the separation in not sufficient to follow the voices, and version (c) is closest to the human performance for moderate voice separation.

Self-motion with Hearing Impairment and (Directional) Hearing Aids

When listening to a sound source in everyday situations, typical movement behavior is highly individual and may not result in the listener directly facing the sound source. Behavioral differences can affect the performance of directional algorithms in hearing aids, as was shown in previous work by using head movement trajectories of normal-hearing (NH) listeners in acoustic simulations for noise-suppression performance predictions. However, the movement behavior of hearing-impaired (HI) listeners with or without hearing aids may differ, and hearing-aid users might adapt their self-motion to improve the performance of directional algorithms. This work investigates the influence of hearing impairment on self-motion, and the interaction of hearing aids with self-motion. In order to do this, the self-motion of three HI participant groups­­­—aided with an adaptive differential microphone (ADM), aided without ADM, and unaided—was measured and compared to previously measured self-motion data from younger and older NH participants. Self-motion was measured in virtual audiovisual environments (VEs) in the laboratory, and the signal-to-noise ratios (SNRs) and SNR improvement of the ADM resulting from the head movements of the participants were estimated using acoustic simulations. HI participants did almost all of the movement with their head and less with their eyes compared to NH participants, which led to a 0.3 dB increase in estimated SNR and to differences in estimated SNR improvement of the ADM. However, the self-motion of the HI participants aided with ADM was similar to that of other HI participants, indicating that the ADM did not cause listeners to adapt their self-motion.

Open community platform for hearing aid algorithm research: open Master Hearing Aid (openMHA)

open Master Hearing Aid (openMHA) was developed and provided to the hearing aid research community as an open-source software platform with the aim to support sustainable and reproducible research towards improvement and new types of assistive hearing systems not limited by proprietary software. The software offers a flexible framework that allows the users to conduct hearing aid research using tools and a number of signal processing plugins provided with the software as well as the implementation of own methods. The openMHA software is independent of a specific hardware and supports Linux, macOS and Windows operating systems as well as 32-bit and 64-bit ARM-based architectures such as used in small portable integrated systems. www.openmha.org.

The Concurrent OLSA Test: A Method for Speech Recognition in Multi-talker Situations at Fixed SNR

A multi-talker paradigm is introduced that uses different attentional processes to adjust speech-recognition scores with the goal of conducting measurements at high signal-to-noise ratios (SNR). The basic idea is to simulate a group conversation with three talkers. Talkers alternately speak sentences of the German matrix test OLSA. Each time a sentence begins with the name “Kerstin” (call sign), the participant is addressed and instructed to repeat the last words of all sentences from that talker, until another talker begins a sentence with “Kerstin”. The alternation of the talkers is implemented with an adjustable overlap time that causes an overlap between the call sign “Kerstin” and the target words to be repeated. Thus, the two tasks of detecting “Kerstin” and repeating target words are to be done at the same time. The paradigm was tested with 22 young normal-hearing participants (YNH) for three overlap times (0.6 s, 0.8 s, 1.0 s). Results for these overlap times show significant differences, with median target word recognition scores of 88%, 82%, and 77%, respectively (including call-sign and dual-task effects). A comparison of the dual task with the corresponding single tasks suggests that the observed effects reflect an increased cognitive load.

A Step towards Neuro-Steered Hearing Aids: Integrated Portable Setup for Time- Synchronized Acoustic Stimuli Presentation and EEG Recording

Aiming to provide a portable research platform to develop algorithms for neuro-steered hearing aids, a joint hearing aid - EEG measurement setup was implemented in this work. The setup combines the miniaturized electroencephalography sensor technology cEEGrid with a portable hearing aid research platform - the Portable Hearing Laboratory. The different components of the system are connected wirelessly, using the lab streaming layer framework for synchronization of audio and EEG data streams. Our setup was shown to be suitable for simultaneous recording of audio and EEG signals used in a pilot study (n=5) to perform an auditory Oddball experiment. The analysis showed that the setup can reliably capture typical event-related potential responses. Furthermore, linear discriminant analysis was successfully applied for single-trial classification of P300 responses. The study showed that time-synchronized audio and EEG data acquisition is possible with the Portable Hearing Laboratory research platform.

The Virtual Reality Lab: Realization and Application of Virtual Sound Environments

To assess perception with and performance of modern and future hearing devices with advanced adaptive signal processing capabilities, novel evaluation methods are required that go beyond already established methods. These novel methods will simulate to a certain extent the complexity and variability of acoustic conditions and acoustic communication styles in real life. This article discusses the current state and the perspectives of virtual reality technology use in the lab for designing complex audiovisual communication environments for hearing assessment and hearing device design and evaluation. In an effort to increase the ecological validity of lab experiments, that is, to increase the degree to which lab data reflect real-life hearing-related function, and to support the development of improved hearing-related procedures and interventions, this virtual reality lab marks a transition from conventional (audio-only) lab experiments to the field. The first part of the article introduces and discusses the notion of the communication loop as a theoretical basis for understanding the factors that are relevant for acoustic communication in real life. From this, requirements are derived that allow an assessment of the extent to which a virtual reality lab reflects these factors, and which may be used as a proxy for ecological validity. The most important factor of real-life communication identified is a closed communication loop among the actively behaving participants. The second part of the article gives an overview of the current developments towards a virtual reality lab at Oldenburg University that aims at interactive and reproducible testing of subjects with and without hearing devices in challenging communication conditions. The extent to which the virtual reality lab in its current state meets the requirements defined in the first part is discussed, along with its limitations and potential further developments. Finally, data are presented from a qualitative study that compared subject behavior and performance in two audiovisual environments presented in the virtual reality lab-a street and a cafeteria-with the corresponding field environments. The results show similarities and differences in subject behavior and performance between the lab and the field, indicating that the virtual reality lab in its current state marks a step towards more ecological validity in lab-based hearing and hearing device research, but requires further development towards higher levels of ecological validity.

The Quest for Ecological Validity in Hearing Science: What It Is, Why It Matters, and How to Advance It

Ecological validity is a relatively new concept in hearing science. It has been cited as relevant with increasing frequency in publications over the past 20 years, but without any formal conceptual basis or clear motive. The sixth Eriksholm Workshop was convened to develop a deeper understanding of the concept for the purpose of applying it in hearing research in a consistent and productive manner. Inspired by relevant debate within the field of psychology, and taking into account the World Health Organization's International Classification of Functioning, Disability, and Health framework, the attendees at the workshop reached a consensus on the following definition: "In hearing science, ecological validity refers to the degree to which research findings reflect real-life hearing-related function, activity, or participation." Four broad purposes for striving for greater ecological validity in hearing research were determined: A (Understanding) better understanding the role of hearing in everyday life; B (Development) supporting the development of improved procedures and interventions; C (Assessment) facilitating improved methods for assessing and predicting ability to accomplish real-world tasks; and D (Integration and Individualization) enabling more integrated and individualized care. Discussions considered the effects of variables and phenomena commonly present in hearing-related research on the level of ecological validity of outcomes, supported by examples from a few selected outcome domains and for different types of studies. Illustrated with examples, potential strategies were offered for promoting a high level of ecological validity in a study and for how to evaluate the level of ecological validity of a study. Areas in particular that could benefit from more research to advance ecological validity in hearing science include: (1) understanding the processes of hearing and communication in everyday listening situations, and specifically the factors that make listening difficult in everyday situations; (2) developing new test paradigms that include more than one person (e.g., to encompass the interactive nature of everyday communication) and that are integrative of other factors that interact with hearing in real-life function; (3) integrating new and emerging technologies (e.g., virtual reality) with established test methods; and (4) identifying the key variables and phenomena affecting the level of ecological validity to develop verifiable ways to increase ecological validity and derive a set of benchmarks to strive for.

Development and evaluation of video recordings for the OLSA matrix sentence test

OBJECTIVE

The aim was to create and validate an audiovisual version of the German matrix sentence test (MST), which uses the existing audio-only speech material.

DESIGN

Video recordings were recorded and dubbed with the audio of the existing German MST. The current study evaluates the MST in conditions including audio and visual modalities, speech in quiet and noise, and open and closed-set response formats.

SAMPLE

One female talker recorded repetitions of the German MST sentences. Twenty-eight young normal-hearing participants completed the evaluation study.

RESULTS

The audiovisual benefit in quiet was 7.0 dB in sound pressure level (SPL). In noise, the audiovisual benefit was 4.9 dB in signal-to-noise ratio (SNR). Speechreading scores ranged from 0% to 84% speech reception in visual-only sentences (mean = 50%). Audiovisual speech reception thresholds (SRTs) had a larger standard deviation than audio-only SRTs. Audiovisual SRTs improved successively with increasing number of lists performed. The final video recordings are openly available.

CONCLUSIONS

The video material achieved similar results as the literature in terms of gross speech intelligibility, despite the inherent asynchronies of dubbing. Due to ceiling effects, adaptive procedures targeting 80% intelligibility should be used. At least one or two training lists should be performed.

Can You Hear Out the Melody? Testing Musical Scene Perception in Young Normal-Hearing and Older Hearing-Impaired Listeners

It is well known that hearing loss compromises auditory scene analysis abilities, as is usually manifested in difficulties of understanding speech in noise. Remarkably little is known about auditory scene analysis of hearing-impaired (HI) listeners when it comes to musical sounds. Specifically, it is unclear to which extent HI listeners are able to hear out a melody or an instrument from a musical mixture. Here, we tested a group of younger normal-hearing (yNH) and older HI (oHI) listeners with moderate hearing loss in their ability to match short melodies and instruments presented as part of mixtures. Four-tone sequences were used in conjunction with a simple musical accompaniment that acted as a masker (cello/piano dyads or spectrally matched noise). In each trial, a signal-masker mixture was presented, followed by two different versions of the signal alone. Listeners indicated which signal version was part of the mixture. Signal versions differed either in terms of the sequential order of the pitch sequence or in terms of timbre (flute vs. trumpet). Signal-to-masker thresholds were measured by varying the signal presentation level in an adaptive two-down/one-up procedure. We observed that thresholds of oHI listeners were elevated by on average 10 dB compared with that of yNH listeners. In contrast to yNH listeners, oHI listeners did not show evidence of listening in dips of the masker. Musical training of participants was associated with a lowering of thresholds. These results may indicate detrimental effects of hearing loss on central aspects of musical scene perception.

Spatial Acoustic Scenarios in Multichannel Loudspeaker Systems for Hearing Aid Evaluation

Background: Field tests and guided walks in real environments show that the benefit from hearing aid (HA) signal processing in real-life situations is typically lower than the predicted benefit found in laboratory studies. This suggests that laboratory test outcome measures are poor predictors of real-life HA benefits. However, a systematic evaluation of algorithms in the field is difficult due to the lack of reproducibility and control of the test conditions. Virtual acoustic environments that simulate real-life situations may allow for a systematic and reproducible evaluation of HAs under more realistic conditions, thus providing a better estimate of real-life benefit than established laboratory tests.

Purpose: To quantify the difference in HA performance between a laboratory condition and more realistic conditions based on technical performance measures using virtual acoustic environments, and to identify the factors affecting HA performance across the tested environments.

Research Design: A set of typical HA beamformer algorithms was evaluated in virtual acoustic environments of different complexity. Performance was assessed based on established technical performance measures, including perceptual model predictions of speech quality and speech intelligibility. Virtual acoustic environments ranged from a simple static reference condition to more realistic complex scenes with dynamically moving sound objects.

Results: HA benefit, as predicted by signal-to-noise ratio (SNR) and speech intelligibility measures, differs between the reference condition and more realistic conditions for the tested beamformer algorithms. Other performance measures, such as speech quality or binaural degree of diffusiveness, do not show pronounced differences. However, a decreased speech quality was found in specific conditions. A correlation analysis showed a significant correlation between room acoustic parameters of the sound field and HA performance. The SNR improvement in the reference condition was found to be a poor predictor of HA performance in terms of speech intelligibility improvement in the more realistic conditions.

Conclusions: Using several virtual acoustic environments of different complexity, a systematic difference in HA performance between a simple reference condition and more realistic environments was found, which may be related to the discrepancy between laboratory and real-life HA performance reported previously.

Evaluation of the Influence of Head Movement on Hearing Aid Algorithm Performance Using Acoustic Simulations

Head movements can improve sound localization performance and speech intelligibility in acoustic environments with spatially distributed sources. However, they can affect the performance of hearing aid algorithms, when adaptive algorithms have to adjust to changes in the acoustic scene caused by head movement (the so-called maladaptation effect) or when directional algorithms are not facing in the optimal direction because the head has moved away (the so-called misalignment effect). In this article, we investigated the mechanisms behind these maladaptation and misalignment effects for a set of six standard hearing aid algorithms using acoustic simulations based on premade databases; this was done so we could study the effects as carefully as possible. Experiment 1 investigated the maladaptation effect by analyzing hearing aid benefit after simulated rotational head movement in simple anechoic noise scenarios. The effects of movement parameters (start angle and peak velocity), noise scenario complexity, and adaptation time were studied, as well as the recovery time of the algorithms. However, a significant maladaptation effect was only found in the most unrealistic anechoic scenario with one noise source. Experiment 2 investigated the effects of maladaptation and misalignment using previously recorded natural head movements in acoustic scenes resembling everyday life situations. In line with the results of Experiment 1, no effect of maladaptation was found in these more realistic acoustic scenes. However, a significant effect of misalignment on the performance of directional algorithms was found. This demonstrates the need to take head movement into account in the evaluation of directional hearing aid algorithms.

Age Effects on Concurrent Speech Segregation by Onset Asynchrony

Purpose For elderly listeners, it is more challenging to listen to 1 voice surrounded by other voices than for young listeners. This could be caused by a reduced ability to use acoustic cues-such as slight differences in onset time-for the segregation of concurrent speech signals. Here, we study whether the ability to benefit from onset asynchrony differs between young (18-33 years) and elderly (55-74 years) listeners. Method We investigated young (normal hearing, N = 20) and elderly (mildly hearing impaired, N = 26) listeners' ability to segregate 2 vowels with onset asynchronies ranging from 20 to 100 ms. Behavioral measures were complemented by a specific event-related brain potential component, the object-related negativity, indicating the perception of 2 distinct auditory objects. Results Elderly listeners' behavioral performance (identification accuracy of the 2 vowels) was considerably poorer than young listeners'. However, both age groups showed the same amount of improvement with increasing onset asynchrony. Object-related negativity amplitude also increased similarly in both age groups. Conclusion Both age groups benefit to a similar extent from onset asynchrony as a cue for concurrent speech segregation during active (behavioral measurement) and during passive (electroencephalographic measurement) listening.

Movement and Gaze Behavior in Virtual Audiovisual Listening Environments Resembling Everyday Life

Recent achievements in hearing aid development, such as visually guided hearing aids, make it increasingly important to study movement behavior in everyday situations in order to develop test methods and evaluate hearing aid performance. In this work, audiovisual virtual environments (VEs) were designed for communication conditions in a living room, a lecture hall, a cafeteria, a train station, and a street environment. Movement behavior (head movement, gaze direction, and torso rotation) and electroencephalography signals were measured in these VEs in the laboratory for 22 younger normal-hearing participants and 19 older normal-hearing participants. These data establish a reference for future studies that will investigate the movement behavior of hearing-impaired listeners and hearing aid users for comparison. Questionnaires were used to evaluate the subjective experience in the VEs. A test-retest comparison showed that the measured movement behavior is reproducible and that the measures of movement behavior used in this study are reliable. Moreover, evaluation of the questionnaires indicated that the VEs are sufficiently realistic. The participants rated the experienced acoustic realism of the VEs positively, and although the rating of the experienced visual realism was lower, the participants felt to some extent present and involved in the VEs. Analysis of the movement data showed that movement behavior depends on the VE and the age of the subject and is predictable in multitalker conversations and for moving distractors. The VEs and a database of the collected data are publicly available.

Effects of directional hearing aid settings on different laboratory measures of spatial awareness perception

Hearing loss can negatively influence the spatial hearing abilities of hearing-impaired listeners, not only in static but also in dynamic auditory environments. Therefore, ways of addressing these deficits with advanced hearing aid algorithms need to be investigated. In a previous study based on virtual acoustics and a computer simulation of different bilateral hearing aid fittings, we investigated auditory source movement detectability in older hearing- impaired (OHI) listeners. We found that two directional processing algorithms could substantially improve the detectability of left-right and near-far source movements in the presence of reverberation and multiple interfering sounds. In the current study, we carried out similar measurements with a loudspeaker-based setup and wearable hearing aids. We fitted a group of 15 OHI listeners with bilateral behind-the-ear devices that were programmed to have three different directional processing settings. Apart from source movement detectability, we assessed two other aspects of spatial awareness perception. Using a street scene with up to five environmental sound sources, the participants had to count the number of presented sources or to indicate the movement direction of a single target signal. The data analyses showed a clear influence of the number of concurrent sound sources and the starting position of the moving target signal on the participants' performance, but no influence of the different hearing aid settings. Complementary artificial head recordings showed that the acoustic differences between the three hearing aid settings were rather small. Another explanation for the lack of effects of the tested hearing aid settings could be that the simulated street scenario was not sufficiently sensitive. Possible ways of improving the sensitivity of the laboratory measures while maintaining high ecological validity and complexity are discussed.

Influence of Multi-microphone Signal Enhancement Algorithms on the Acoustics and Detectability of Angular and Radial Source Movements

Hearing-impaired listeners are known to have difficulties not only with understanding speech in noise but also with judging source distance and movement, and these deficits are related to perceived handicap. It is possible that the perception of spatially dynamic sounds can be improved with hearing aids (HAs), but so far this has not been investigated. In a previous study, older hearing-impaired listeners showed poorer detectability for virtual left-right (angular) and near-far (radial) source movements due to lateral interfering sounds and reverberation, respectively. In the current study, potential ways of improving these deficits with HAs were explored. Using stimuli very similar to before, detailed acoustic analyses were carried out to examine the influence of different HA algorithms for suppressing noise and reverberation on the acoustic cues previously shown to be associated with source movement detectability. For an algorithm that combined unilateral directional microphones with binaural coherence-based noise reduction and for a bilateral beamformer with binaural cue preservation, movement-induced changes in spectral coloration, signal-to-noise ratio, and direct-to-reverberant energy ratio were greater compared with no HA processing. To evaluate these two algorithms perceptually, aided measurements of angular and radial source movement detectability were performed with 20 older hearing-impaired listeners. The analyses showed that, in the presence of concurrent interfering sounds and reverberation, the bilateral beamformer could restore source movement detectability in both spatial dimensions, whereas the other algorithm only improved detectability in the near-far dimension. Together, these results provide a basis for improving the detectability of spatially dynamic sounds with HAs.

Sparse periodicity-based auditory features explain human performance in a spatial multitalker auditory scene analysis task

Human listeners robustly decode speech information from a talker of interest that is embedded in a mixture of spatially distributed interferers. A relevant question is which time-frequency segments of the speech are predominantly used by a listener to solve such a complex Auditory Scene Analysis task. A recent psychoacoustic study investigated the relevance of low signal-to-noise ratio (SNR) components of a target signal on speech intelligibility in a spatial multitalker situation. For this, a three-talker stimulus was manipulated in the spectro-temporal domain such that target speech time-frequency units below a variable SNR threshold (SNR_crit ) were discarded while keeping the interferers unchanged. The psychoacoustic data indicate that only target components at and above a local SNR of about 0 dB contribute to intelligibility. This study applies an auditory scene analysis "glimpsing" model to the same manipulated stimuli. Model data are found to be similar to the human data, supporting the notion of "glimpsing," that is, that salient speech-related information is predominantly used by the auditory system to decode speech embedded in a mixture of sounds, at least for the tested conditions of three overlapping speech signals. This implies that perceptually relevant auditory information is sparse and may be processed with low computational effort, which is relevant for neurophysiological research of scene analysis and novelty processing in the auditory system.

The effects of electrical field spatial spread and some cognitive factors on speech-in-noise performance of individual cochlear implant users-A computer model study

The relation of the individual speech-in-noise performance differences in cochlear implant (CI) users to underlying physiological factors is currently poorly understood. This study approached this research question by a step-wise individualization of a computer model of speech intelligibility mimicking the details of CI signal processing and some details of the physiology present in CI users. Two factors, the electrical field spatial spread and internal noise (as a coarse model of the individual cognitive performance) were incorporated. Internal representations of speech-in-noise mixtures calculated by the model were classified using an automatic speech recognizer backend employing Hidden Markov Models with a Gaussian probability distribution. One-dimensional electric field spatial spread functions were inferred from electrical field imaging data of 14 CI users. Simplified assumptions of homogenously distributed auditory nerve fibers along the cochlear array and equal distance between electrode array and nerve tissue were assumed in the model. Internal noise, whose standard deviation was adjusted based on either anamnesis data, or text-reception-threshold data, or a combination thereof, was applied to the internal representations before classification. A systematic model evaluation showed that predicted speech-reception-thresholds (SRTs) in stationary noise improved (decreased) with decreasing internal noise standard deviation and with narrower electric field spatial spreads. The model version that was individualized to actual listeners using internal noise alone (containing average spatial spread) showed significant correlations to measured SRTs, reflecting the high correlation of the text-reception threshold data with SRTs. However, neither individualization to spatial spread functions alone, nor a combined individualization based on spatial spread functions and internal noise standard deviation did produce significant correlations with measured SRTs.

Combination of binaural and harmonic masking release effects in the detection of a single component in complex tones

Both harmonic and binaural signal properties are relevant for auditory processing. To investigate how these cues combine in the auditory system, detection thresholds for an 800-Hz tone masked by a diotic (i.e., identical between the ears) harmonic complex tone were measured in six normal-hearing subjects. The target tone was presented either diotically or with an interaural phase difference (IPD) of 180° and in either harmonic or "mistuned" relationship to the diotic masker. Three different maskers were used, a resolved and an unresolved complex tone (fundamental frequency: 160 and 40 Hz) with four components below and above the target frequency and a broadband unresolved complex tone with 12 additional components. The target IPD provided release from masking in most masker conditions, whereas mistuning led to a significant release from masking only in the diotic conditions with the resolved and the narrowband unresolved maskers. A significant effect of mistuning was neither found in the diotic condition with the wideband unresolved masker nor in any of the dichotic conditions. An auditory model with a single analysis frequency band and different binaural processing schemes was employed to predict the data of the unresolved masker conditions. Sensitivity to modulation cues was achieved by including an auditory-motivated modulation filter in the processing pathway. The predictions of the diotic data were in line with the experimental results and literature data in the narrowband condition, but not in the broadband condition, suggesting that across-frequency processing is involved in processing modulation information. The experimental and model results in the dichotic conditions show that the binaural processor cannot exploit modulation information in binaurally unmasked conditions.

Modeling speech localization, talker identification, and word recognition in a multi-talker setting

This study introduces a model for solving three different auditory tasks in a multi-talker setting: target localization, target identification, and word recognition. The model was used to simulate psychoacoustic data from a call-sign-based listening test involving multiple spatially separated talkers [Brungart and Simpson (2007). Percept. Psychophys. 69(1), 79-91]. The main characteristics of the model are (i) the extraction of salient auditory features ("glimpses") from the multi-talker signal and (ii) the use of a classification method that finds the best target hypothesis by comparing feature templates from clean target signals to the glimpses derived from the multi-talker mixture. The four features used were periodicity, periodic energy, and periodicity-based interaural time and level differences. The model results widely exceeded probability of chance for all subtasks and conditions, and generally coincided strongly with the subject data. This indicates that, despite their sparsity, glimpses provide sufficient information about a complex auditory scene. This also suggests that complex source superposition models may not be needed for auditory scene analysis. Instead, simple models of clean speech may be sufficient to decode even complex multi-talker scenes.

Evaluation of combined dynamic compression and single channel noise reduction for hearing aid applications

OBJECTIVE

Single-channel noise reduction (SCNR) and dynamic range compression (DRC) are important elements in hearing aids. Only relatively few studies have addressed interaction effects and typically used real hearing aids with limited knowledge about the integrated algorithms. Here the potential benefit of different combinations and integration of SCNR and DRC was systematically assessed.

DESIGN

Ten different systems combining SCNR and DRC were implemented, including five serial arrangements, a parallel and two multiplicative approaches. In an instrumental evaluation, signal-to-noise ratio (SNR) improvement and spectral contrast enhancement (SCE) were assessed. Quality ratings at 0 and +6 dB SNR, and speech reception thresholds (SRTs) in noise were measured using stationary and babble noise.

STUDY SAMPLE

Thirteen young normal-hearing (NH) listeners and 12 hearing-impaired (HI) listeners participated.

RESULTS

In line with an increased segmental SNR and spectral contrast compared to a serial concatenation, the parallel approach significantly reduced the perceived noise annoyance for both subject groups. The proposed multiplicative approaches could partly counteract increased speech distortions introduced by DRC and achieved the best overall quality for the HI listeners.

CONCLUSIONS

For high SNRs well above the individual SRT, the specific combination of SCNR and DRC is perceptually relevant and the integrative approaches were preferred.

Sensitivity to Angular and Radial Source Movements as a Function of Acoustic Complexity in Normal and Impaired Hearing

In contrast to static sounds, spatially dynamic sounds have received little attention in psychoacoustic research so far. This holds true especially for acoustically complex (reverberant, multisource) conditions and impaired hearing. The current study therefore investigated the influence of reverberation and the number of concurrent sound sources on source movement detection in young normal-hearing (YNH) and elderly hearing-impaired (EHI) listeners. A listening environment based on natural environmental sounds was simulated using virtual acoustics and rendered over headphones. Both near-far ('radial') and left-right ('angular') movements of a frontal target source were considered. The acoustic complexity was varied by adding static lateral distractor sound sources as well as reverberation. Acoustic analyses confirmed the expected changes in stimulus features that are thought to underlie radial and angular source movements under anechoic conditions and suggested a special role of monaural spectral changes under reverberant conditions. Analyses of the detection thresholds showed that, with the exception of the single-source scenarios, the EHI group was less sensitive to source movements than the YNH group, despite adequate stimulus audibility. Adding static sound sources clearly impaired the detectability of angular source movements for the EHI (but not the YNH) group. Reverberation, on the other hand, clearly impaired radial source movement detection for the EHI (but not the YNH) listeners. These results illustrate the feasibility of studying factors related to auditory movement perception with the help of the developed test setup.

Virtual acoustic environments for comprehensive evaluation of model-based hearing devices<sup/>

OBJECTIVE

Create virtual acoustic environments (VAEs) with interactive dynamic rendering for applications in audiology.

DESIGN

A toolbox for creation and rendering of dynamic virtual acoustic environments (TASCAR) that allows direct user interaction was developed for application in hearing aid research and audiology. The software architecture and the simulation methods used to produce VAEs are outlined. Example environments are described and analysed.

CONCLUSION

With the proposed software, a tool for simulation of VAEs is available. A set of VAEs rendered with the proposed software was described.

Modeling of speech localization in a multi-talker mixture using periodicity and energy-based auditory features

A recent study showed that human listeners are able to localize a short speech target simultaneously masked by four speech tokens in reverberation [Kopčo, Best, and Carlile (2010). J. Acoust. Soc. Am. 127, 1450-1457]. Here, an auditory model for solving this task is introduced. The model has three processing stages: (1) extraction of the instantaneous interaural time difference (ITD) information, (2) selection of target-related ITD information ("glimpses") using a template-matching procedure based on periodicity, spectral energy, or both, and (3) target location estimation. The model performance was compared to the human data, and to the performance of a modified model using an ideal binary mask (IBM) at stage (2). The IBM-based model performed similarly to the subjects, indicating that the binaural model is able to accurately estimate source locations. Template matching using spectral energy and using a combination of spectral energy and periodicity achieved good results, while using periodicity alone led to poor results. Particularly, the glimpses extracted from the initial portion of the signal were critical for good performance. Simulation data show that the auditory features investigated here are sufficient to explain human performance in this challenging listening condition and thus may be used in models of auditory scene analysis.

Spectral and binaural loudness summation for hearing-impaired listeners

Sensorineural hearing loss typically results in a steepened loudness function and a reduced dynamic range from elevated thresholds to uncomfortably loud levels for narrowband and broadband signals. Restoring narrowband loudness perception for hearing-impaired (HI) listeners can lead to overly loud perception of broadband signals and it is unclear how binaural presentation affects loudness perception in this case. Here, loudness perception quantified by categorical loudness scaling for nine normal-hearing (NH) and ten HI listeners was compared for signals with different bandwidth and different spectral shape in monaural and in binaural conditions. For the HI listeners, frequency- and level-dependent amplification was used to match the narrowband monaural loudness functions of the NH listeners. The average loudness functions for NH and HI listeners showed good agreement for monaural broadband signals. However, HI listeners showed substantially greater loudness for binaural broadband signals than NH listeners: on average a 14.1 dB lower level was required to reach "very loud" (range 30.8 to -3.7 dB). Overall, with narrowband loudness compensation, a given binaural loudness for broadband signals above "medium loud" was reached at systematically lower levels for HI than for NH listeners. Such increased binaural loudness summation was not found for loudness categories below "medium loud" or for narrowband signals. Large individual variations in the increased loudness summation were observed and could not be explained by the audiogram or the narrowband loudness functions.

Spatial Release From Masking in Simulated Cochlear Implant Users With and Without Access to Low-Frequency Acoustic Hearing

For normal-hearing listeners, speech intelligibility improves if speech and noise are spatially separated. While this spatial release from masking has already been quantified in normal-hearing listeners in many studies, it is less clear how spatial release from masking changes in cochlear implant listeners with and without access to low-frequency acoustic hearing. Spatial release from masking depends on differences in access to speech cues due to hearing status and hearing device. To investigate the influence of these factors on speech intelligibility, the present study measured speech reception thresholds in spatially separated speech and noise for 10 different listener types. A vocoder was used to simulate cochlear implant processing and low-frequency filtering was used to simulate residual low-frequency hearing. These forms of processing were combined to simulate cochlear implant listening, listening based on low-frequency residual hearing, and combinations thereof. Simulated cochlear implant users with additional low-frequency acoustic hearing showed better speech intelligibility in noise than simulated cochlear implant users without acoustic hearing and had access to more spatial speech cues (e.g., higher binaural squelch). Cochlear implant listener types showed higher spatial release from masking with bilateral access to low-frequency acoustic hearing than without. A binaural speech intelligibility model with normal binaural processing showed overall good agreement with measured speech reception thresholds, spatial release from masking, and spatial speech cues. This indicates that differences in speech cues available to listener types are sufficient to explain the changes of spatial release from masking across these simulated listener types.

Comparing Binaural Pre-processing Strategies II: Speech Intelligibility of Bilateral Cochlear Implant Users

Several binaural audio signal enhancement algorithms were evaluated with respect to their potential to improve speech intelligibility in noise for users of bilateral cochlear implants (CIs). 50% speech reception thresholds (SRT₅₀) were assessed using an adaptive procedure in three distinct, realistic noise scenarios. All scenarios were highly nonstationary, complex, and included a significant amount of reverberation. Other aspects, such as the perfectly frontal target position, were idealized laboratory settings, allowing the algorithms to perform better than in corresponding real-world conditions. Eight bilaterally implanted CI users, wearing devices from three manufacturers, participated in the study. In all noise conditions, a substantial improvement in SRT₅₀ compared to the unprocessed signal was observed for most of the algorithms tested, with the largest improvements generally provided by binaural minimum variance distortionless response (MVDR) beamforming algorithms. The largest overall improvement in speech intelligibility was achieved by an adaptive binaural MVDR in a spatially separated, single competing talker noise scenario. A no-pre-processing condition and adaptive differential microphones without a binaural link served as the two baseline conditions. SRT₅₀ improvements provided by the binaural MVDR beamformers surpassed the performance of the adaptive differential microphones in most cases. Speech intelligibility improvements predicted by instrumental measures were shown to account for some but not all aspects of the perceptually obtained SRT₅₀ improvements measured in bilaterally implanted CI users.

Comparing Binaural Pre-processing Strategies I: Instrumental Evaluation

In a collaborative research project, several monaural and binaural noise reduction algorithms have been comprehensively evaluated. In this article, eight selected noise reduction algorithms were assessed using instrumental measures, with a focus on the instrumental evaluation of speech intelligibility. Four distinct, reverberant scenarios were created to reflect everyday listening situations: a stationary speech-shaped noise, a multitalker babble noise, a single interfering talker, and a realistic cafeteria noise. Three instrumental measures were employed to assess predicted speech intelligibility and predicted sound quality: the intelligibility-weighted signal-to-noise ratio, the short-time objective intelligibility measure, and the perceptual evaluation of speech quality. The results show substantial improvements in predicted speech intelligibility as well as sound quality for the proposed algorithms. The evaluated coherence-based noise reduction algorithm was able to provide improvements in predicted audio signal quality. For the tested single-channel noise reduction algorithm, improvements in intelligibility-weighted signal-to-noise ratio were observed in all but the nonstationary cafeteria ambient noise scenario. Binaural minimum variance distortionless response beamforming algorithms performed particularly well in all noise scenarios.

A Binaural Steering Beamformer System for Enhancing a Moving Speech Source

In many daily life communication situations, several sound sources are simultaneously active. While normal-hearing listeners can easily distinguish the target sound source from interfering sound sources-as long as target and interferers are spatially or spectrally separated-and concentrate on the target, hearing-impaired listeners and cochlear implant users have difficulties in making such a distinction. In this article, we propose a binaural approach composed of a spatial filter controlled by a direction-of-arrival estimator to track and enhance a moving target sound. This approach was implemented on a real-time signal processing platform enabling experiments with test subjects in situ. To evaluate the proposed method, a data set of sound signals with a single moving sound source in an anechoic diffuse noise environment was generated using virtual acoustics. The proposed steering method was compared with a fixed (nonsteering) method that enhances sound from the frontal direction in an objective evaluation and subjective experiments using this database. In both cases, the obtained results indicated a significant improvement in speech intelligibility and quality compared with the unprocessed signal. Furthermore, the proposed method outperformed the nonsteering method.

Robust auditory localization using probabilistic inference and coherence-based weighting of interaural cues

Robust sound source localization is performed by the human auditory system even in challenging acoustic conditions and in previously unencountered, complex scenarios. Here a computational binaural localization model is proposed that possesses mechanisms for handling of corrupted or unreliable localization cues and generalization across different acoustic situations. Central to the model is the use of interaural coherence, measured as interaural vector strength (IVS), to dynamically weight the importance of observed interaural phase (IPD) and level (ILD) differences in frequency bands up to 1.4 kHz. This is accomplished through formulation of a probabilistic model in which the ILD and IPD distributions pertaining to a specific source location are dependent on observed interaural coherence. Bayesian computation of the direction-of-arrival probability map naturally leads to coherence-weighted integration of location cues across frequency and time. Results confirm the model's validity through statistical analyses of interaural parameter values. Simulated localization experiments show that even data points with low reliability (i.e., low IVS) can be exploited to enhance localization performance. A temporal integration length of at least 200 ms is required to gain a benefit; this is in accordance with previous psychoacoustic findings on temporal integration of spatial cues in the human auditory system.

The influence of pause, attack, and decay duration of the ongoing envelope on sound lateralization

Klein-Hennig et al. [J. Acoust. Soc. Am. 129, 3856-3872 (2011)] introduced a class of high-frequency stimuli for which the envelope shape can be altered by independently varying the attack, hold, decay, and pause durations. These stimuli, originally employed for testing the shape dependence of human listeners' sensitivity to interaural temporal differences (ITDs) in the ongoing envelope, were used to measure the lateralization produced by fixed interaural disparities. Consistent with the threshold ITD data, a steep attack and a non-zero pause facilitate strong ITD-based lateralization. In contrast, those conditions resulted in the smallest interaural level-based lateralization.

The effect of overall level on sensitivity to interaural differences of time and level at high frequencies

For high-frequency complex stimuli, detection thresholds for envelope-based interaural time differences (ITDs) decrease with overall level. Substantial heterogeneity is, however, evident among the findings concerning the rate at which thresholds decline with level. This study investigated factors affecting the influence of overall level on threshold ITDs. Thresholds were measured as a function of overall level for 4-kHz-centered "targets" in three experiments focusing, respectively, on stimulus-type (sinusoidally amplitude-modulated or "transposed" tones), modulation frequency, and details concerning low-pass noise used to mask low-frequency distortion products. Results indicated that (1) log-ITD thresholds decreased linearly with overall level; (2) slopes relating log-ITD thresholds to level did not depend significantly on stimulus type; (3) lower modulation frequencies produced greater dependencies of thresholds on overall level than did higher modulation frequencies; (4) the effect of overall level on threshold-ITDs was independent of the interaural configuration and levels of the low-pass noise maskers tested; (5) synchronously gating the low-pass noise and target produced a greater dependency of thresholds on the overall level of the target than did continuous or temporally "fringed" presentation of the noise. A fourth experiment showed that threshold interaural level differences were somewhat less affected by changes in overall level than were threshold ITDs.

Relation between loudness in categorical units and loudness in phons and sones

Data are presented on the relation between loudness measured in categorical units (CUs) using a standardized loudness scaling method (ISO 16832, 2006) and loudness expressed as the classical standardized measures phon and sone. Based on loudness scaling of narrowband noise signals by 31 normal-hearing subjects, sound pressure levels eliciting the same categorical loudness were derived for various center frequencies. The results were comparable to the standardized equal-loudness level contours. A comparison between the loudness function in CUs at 1000 Hz and the standardized loudness function in sones indicates a cubic relation between the two loudness measures.

Comparing the effect of pause duration on threshold interaural time differences between exponential and squared-sine envelopes (L)

Recently two studies [Klein-Hennig et al., J. Acoust. Soc. Am. 129, 3856-3872 (2011); Laback et al., J. Acoust. Soc. Am. 130, 1515-1529 (2011)] independently investigated the isolated effect of pause duration on sensitivity to interaural time differences (ITD) in the ongoing stimulus envelope. The steepness of the threshold ITD as a function of pause duration functions differed considerably across studies. The present study, using matched carrier and modulation frequencies, directly compared threshold ITDs for the two envelope flank shapes from those studies. The results agree well when defining the metric of pause duration based on modulation depth sensitivity.

Release from masking of low-frequency complex tones by high-frequency complex tone cue bands

This study investigated the influence of high-frequency cue bands on the detection and discrimination of low-frequency target bands presented in a 3000-Hz low-pass noise masker. Target and cue bands were complex tones with 80-Hz spacing. The cue band consisted of 60 components starting at 4000 Hz; targets consisted of four components starting at different frequencies (500, 700, 1000, 1200, and 1500 Hz). Targets were presented with different durations within the 500-ms masker; target and cue bands had a common on- and offset. Presentation of the high-frequency complex tone significantly enhanced both the discrimination and detection thresholds by 2-3 dB.

Effect of mistuning on the detection of a tone masked by a harmonic tone complex

The human auditory system is sensitive in detecting “mistuned” components in a harmonic complex, which do not match the frequency pattern defined by the fundamental frequency of the complex. Depending on the frequency configuration, the mistuned component may be perceptually segregated from the complex and may be heard as a separate tone. In the context of a masking experiment, mistuning a single component decreases its masked threshold. In this study we propose to quantify the ability to detect a single component for fixed amounts of mistuning by adaptively varying its level. This method produces masking release by mistuning that can be compared to other masking release effects. Detection thresholds were obtained for various frequency configurations where the target component was resolved or unresolved in the auditory system. The results from 6 normal-hearing listeners show a significant decrease of masked thresholds between harmonic and mistuned conditions in all configurations and provide evidence for the employment of different detection strategies for resolved and unresolved components. The data suggest that across-frequency processing is involved in the release from masking. The results emphasize the ability of this method to assess integrative aspects of pitch and harmonicity perception.

Evaluation of model-based versus non-parametric monaural noise-reduction approaches for hearing aids

OBJECTIVE

Single channel noise reduction has been well investigated and seems to have reached its limits in terms of speech intelligibility improvement, however, the quality of such schemes can still be advanced. This study tests to what extent novel model-based processing schemes might improve performance in particular for non-stationary noise conditions.

DESIGN

Two prototype model-based algorithms, a speech-model-based, and a auditory-model-based algorithm were compared to a state-of-the-art non-parametric minimum statistics algorithm. A speech intelligibility test, preference rating, and listening effort scaling were performed. Additionally, three objective quality measures for the signal, background, and overall distortions were applied. For a better comparison of all algorithms, particular attention was given to the usage of the similar Wiener-based gain rule.

STUDY SAMPLE

The perceptual investigation was performed with fourteen hearing-impaired subjects.

RESULTS

The results revealed that the non-parametric algorithm and the auditory model-based algorithm did not affect speech intelligibility, whereas the speech-model-based algorithm slightly decreased intelligibility. In terms of subjective quality, both model-based algorithms perform better than the unprocessed condition and the reference in particular for highly non-stationary noise environments.

CONCLUSION

Data support the hypothesis that model-based algorithms are promising for improving performance in non-stationary noise conditions.

Indication criteria for cochlear implants and hearing aids: impact of audiological and non-audiological findings

Owing to technological progress and a growing body of clinical experience, indication criteria for cochlear implants (CI) are being extended to less severe hearing impairments. It is, therefore, worth reconsidering these indication criteria by introducing novel testing procedures. The diagnostic evidence collected will be evaluated. The investigation includes postlingually deafened adults seeking a CI. Prior to surgery, speech perception tests [Freiburg Speech Test and Oldenburg sentence (OLSA) test] were performed unaided and aided using the Oldenburg Master Hearing Aid (MHA) system. Linguistic skills were assessed with the visual Text Reception Threshold (TRT) test, and general state of health, socio-economic status (SES) and subjective hearing were evaluated through questionnaires. After surgery, the speech tests were repeated aided with a CI. To date, 97 complete data sets are available for evaluation. Statistical analyses showed significant correlations between postsurgical speech reception threshold (SRT) measured with the adaptive OLSA test and pre-surgical data such as the TRT test (r=-0.29), SES (r=-0.22) and (if available) aided SRT (r=0.53). The results suggest that new measures and setups such as the TRT test, SES and speech perception with the MHA provide valuable extra information regarding indication for CI.

Lateralization based on interaural differences in the second-order amplitude modulator

Second-order amplitude modulation is a relatively slow variation of the modulation depth of a first-order amplitude modulation with higher frequency. In contrast to first-order modulation, which appears as a physical component in the stimulus spectrum after half-wave rectification, second-order modulation is not necessarily demodulated by the auditory periphery. For binaural processing of second-order amplitude modulated stimuli it is unknown whether interaural time differences (ITDs) in the second-order modulation result in a lateralized percept. Thus, second-order modulation can serve as a tool to investigate whether demodulation of interaurally delayed components is a prerequisite for lateralization. In most of the psychoacoustic experiments presented here, a 25 Hz sinusoidally amplitude-modulated (SAM) 160 Hz tone was either transposed to 4 kHz by half-wave rectifying this SAM waveform before multiplication with a 4 kHz tone (TSAM), or by adding an offset before multiplication (SAMAM). The experiments revealed an inability to lateralize the SAMAM based on ITDs in the 25 Hz component, whereas subjects could lateralize the TSAM. Given that only the TSAM results in a demodulated 25 Hz component after peripheral auditory processing, this result supports the hypothesis that demodulation is a prerequisite for lateralization, which has consequences for temporal modulation processing in models of binaural interaction.

Prevention of Venous Thromboembolism Using Enoxaparin in Day Surgery

We aimed to confirm the results of randomized, controlled trials on enoxaparin prophylaxis in unselected patients undergoing day surgery. The primary end point was the incidence of thromboembolic events during prophylaxis and up to 48 hours thereafter. A total of 11 794 patients, consisting of 52.1% male with mean age of 49.2 ± 15.7 were included. In all, 61.5% had no predisposing risk factors and 67.1% received no concomitant medication with the potential to increase bleeding. Patients were exposed to 20 mg (63.6%) and 40 mg (36.4%) of enoxaparin for a mean of 12.4 ± 9.8 days. Forty-four patients (0.39%) had confirmed symptomatic deep venous thrombosis and 1 patient confirmed pulmonary embolism. Bleeding occurred in 3.47% of patients (3.29% minor bleeding). Differences between 20 and 40 mg enoxaparin were negligible. Adverse drug reactions were experienced by 3.1% of patients. The present study results demonstrate that it is effective and tolerable to use a risk stratified dose of 20 or 40 mg enoxaparin in patients undergoing day surgery.