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 (SNRcrit ) 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.

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.

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.

Virtual acoustic environments for comprehensive evaluation of model-based hearing devices.

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.

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.

A physiologically-inspired model reproducing the speech intelligibility benefit in cochlear implant listeners with residual acoustic hearing.

This study introduces a speech intelligibility model for cochlear implant users with ipsilateral preserved acoustic hearing that aims at simulating the observed speech-in-noise intelligibility benefit when receiving simultaneous electric and acoustic stimulation (EA-benefit). The model simulates the auditory nerve spiking in response to electric and/or acoustic stimulation. The temporally and spatially integrated spiking patterns were used as the final internal representation of noisy speech. Speech reception thresholds (SRTs) in stationary noise were predicted for a sentence test using an automatic speech recognition framework. The model was employed to systematically investigate the effect of three physiologically relevant model factors on simulated SRTs: (1) the spatial spread of the electric field which co-varies with the number of electrically stimulated auditory nerves, (2) the "internal" noise simulating the deprivation of auditory system, and (3) the upper bound frequency limit of acoustic hearing. The model results show that the simulated SRTs increase monotonically with increasing spatial spread for fixed internal noise, and also increase with increasing the internal noise strength for a fixed spatial spread. The predicted EA-benefit does not follow such a systematic trend and depends on the specific combination of the model parameters. Beyond 300 Hz, the upper bound limit for preserved acoustic hearing is less influential on speech intelligibility of EA-listeners in stationary noise. The proposed model-predicted EA-benefits are within the range of EA-benefits shown by 18 out of 21 actual cochlear implant listeners with preserved acoustic hearing.

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.

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.

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.

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.

The influence of different segments of the ongoing envelope on sensitivity to interaural time delays.

The auditory system is sensitive to interaural timing disparities in the fine structure and the envelope of sounds, each contributing important cues for lateralization. In this study, psychophysical measurements were conducted with customized envelope waveforms in order to investigate the isolated effect of different segments of a periodic, ongoing envelope on lateralization. One envelope cycle was composed of the four segments attack flank, hold duration, decay flank, and pause duration, which were independently varied to customize the envelope waveform. The envelope waveforms were applied to a 4-kHz sinusoidal carrier, and just noticeable envelope interaural time differences were measured in six normal hearing subjects. The results indicate that attack durations and pause durations prior to the attack are the most important stimulus characteristics for processing envelope timing disparities. The results were compared to predictions of three binaural lateralization models based on the normalized cross correlation coefficient. Two of the models included an additional stage to mimic neural adaptation prior to binaural interaction, involving either a single short time constant (5 ms) or a combination of five time constants up to 500 ms. It was shown that the model with the single short time constant accounted best for the data.

Sound source localization in real sound fields based on empirical statistics of interaural parameters.

The role of temporal fluctuations and systematic variations of interaural parameters in localization of sound sources in spatially distributed, nonstationary noise conditions was investigated. For this, Bayesian estimation was applied to interaural parameters calculated with physiologically plausible time and frequency resolution. Probability density functions (PDFs) of the interaural level differences (ILDs) and phase differences (IPDs) were estimated by measuring histograms for a directional sound source perturbed by several types of interfering noise at signal-to-noise ratios (SNRs) between -5 and +30 dB. A moment analysis of the PDFs reveals that the expected values shift and the standard deviations increase considerably with decreasing SNR, and that the PDFs have non-Gaussian shape at medium SNRs. A d' analysis of the PDFs indicates that elevation discrimination is possible even at low SNRs in the median plane by integrating information across frequency. Absolute sound localization was simulated by a Bayesian maximum a posteriori (MAP) procedure. The simulation is based on frequency integration of broadly tuned "detectors." Confusion patterns of real and estimated sound source directions are similar to those of human listeners. The results indicate that robust processing strategies are needed to exploit interaural parameters successfully in noise conditions due to their strong temporal fluctuations.