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B Kollmeier

Publications and source records attributed to B Kollmeier.

At least 19 recordsLinked to original sources
PubMed2006

[Evaluation of the Oldenburg children's rhyme test in silence and in noise].

BACKGROUND: The Oldenburg children's rhyme test (OlKi) was designed and optimized for speech intelligibility measurements for primary school pupils in silence [8, 3]. In the optimization, the intelligibility of the particular test words was equalized. METHODS: The evaluation of the test with 147 primary school pupils with normal hearing in silence and 107 pupils in noise is presented in this article. The comparability between test lists for speech intelligibility was investigated and age dependent reference functions were determined. RESULTS: The evaluation showed that intelligibility differences are larger across children within one grade than across different test lists. The reference functions of first grade pupils are shifted to slightly higher presentation levels and signal-to-noise ratios both in silence and noise. CONCLUSIONS: The 12 optimized test lists of the OlKi test are equally intelligible both in silence and noise. No list effects are expected. The degree of difficulty of the Oldenburg children's rhyme test can be compared with the Göttingen children's test II and the Mainz children's test III.

Auditory Perceptual Disorders↗
PubMed2001

Modeling temporal and compressive properties of the normal and impaired auditory system.

Three modifications of a psychoacoustically and physiologically motivated processing model [Dau et al., J. Acoust. Soc. Am. 102 (1997a) 2892-2905] are presented and tested. The modifications aim at simulating sensorineural hearing loss and incorporate a level-dependent peripheral compression whose properties are affected by hearing impairment. Model 1 realizes this difference by introducing for impaired listeners an instantaneous level-dependent expansion prior to the adaptation stage of the model. Model 2 and Model 3 realize a level-dependent compression with time constants of 5 and 15 ms, respectively, for normal hearing and a reduced compression for impaired hearing. In Model 2, the compression occurs after the envelope extraction stage, while in Model 3, envelope extraction follows compression. All models account to a similar extent for the recruitment phenomenon measured with narrow-band stimuli and for forward-masking data of normal-hearing and hearing-impaired subjects using a 20-ms, 2-kHz tone signal and a 1-kHz-wide bandpass noise masker centered at 2 kHz. A clear difference between the different models occurs for the processing of temporally fluctuating stimuli. A modulation-rate-independent increase in modulation-response level for simulating impaired hearing is only predicted by Model 1 while the other two models realize a modulation-rate-dependent increase. Hence, the predictions of Model 2 and Model 3 are in conflict with the results of modulation-matching experiments reported in the literature. It is concluded that key properties of sensorineural hearing loss (altered loudness perception, reduced dynamic range, normal temporal properties but prolonged forward-masking effects) can effectively be modeled by incorporating a fast-acting expansion within the current processing model prior to the nonlinear adaptation stage. Based on these findings, a model of both normal and impaired hearing is proposed which incorporates a fast-acting compressive nonlinearity, representing the cochlear nonlinearity (which is reduced in impaired listeners), followed by an instantaneous expansion and the nonlinear adaptation stage which represent aspects of the retro-cochlear information processing in the auditory system.

Auditory Pathways↗
PubMed2000

Auditory brainstem responses with optimized chirp signals compensating basilar-membrane dispersion.

This study examines auditory brainstem responses (ABR) elicited by rising frequency chirps. The time course of frequency change for the chirp theoretically produces simultaneous displacement maxima by compensating for travel-time differences along the cochlear partition. This broadband chirp was derived on the basis of a linear cochlea model [de Boer, "Auditory physics. Physical principles in hearing theory I," Phys. Rep. 62, 87-174 (1980)]. Responses elicited by the broadband chirp show a larger wave-V amplitude than do click-evoked responses for most stimulation levels tested. This result is in contrast to the general hypothesis that the ABR is an electrophysiological event most effectively evoked by the onset or offset of an acoustic stimulus, and unaffected by further stimulation. The use of this rising frequency chirp enables the inclusion of activity from lower frequency regions, whereas with a click, synchrony is decreased in accordance with decreasing traveling velocity in the apical region. The use of a temporally reversed (falling) chirp leads to a further decrease in synchrony as reflected in ABR responses that are smaller than those from a click. These results are compatible with earlier experimental results from recordings of compound action potentials (CAP) [Shore and Nuttall, "High synchrony compound action potentials evoked by rising frequency-swept tonebursts," J. Acoust. Soc. Am. 78, 1286-1295 (1985)] reflecting activity at the level of the auditory nerve. Since the ABR components considered here presumably reflect neural response from the brainstem, the effect of an optimized synchronization at the peripheral level can also be observed at the brainstem level. The rising chirp may therefore be of clinical use in assessing the integrity of the entire peripheral organ and not just its basal end.

Adult↗
PubMed1999

A model of auditory perception as front end for automatic speech recognition.

A front end for automatic speech recognizers is proposed and evaluated which is based on a quantitative model of the "effective" peripheral auditory processing. The model simulates both spectral and temporal properties of sound processing in the auditory system which were found in psychoacoustical and physiological experiments. The robustness of the auditory-based representation of speech was evaluated in speaker-independent, isolated word recognition experiments in different types of additive noise. The results show a higher robustness of the auditory front end in noise, compared to common mel-scale cepstral feature extraction. In a second set of experiments, different processing stages of the auditory front end were modified to study their contribution to robust speech signal representation in detail. The adaptive compression stage which enhances temporal changes of the input signal appeared to be the most important processing stage towards robust speech representation in noise. Low-pass filtering of the fast fluctuating envelope in each frequency band further reduces the influence of noise in the auditory-based representation of speech.

Auditory Perception↗
PubMed1999

Within-channel cues in comodulation masking release (CMR): experiments and model predictions using a modulation-filterbank model.

Experiments and model calculations were performed to study the influence of within-channel cues versus across-channel cues in comodulation masking release (CMR). A class of CMR experiments is considered that are characterized by a single (unmodulated or modulated) bandpass noise masker with variable bandwidth centered at the signal frequency. A modulation-filterbank model suggested by Dau et al. [J. Acoust. Soc. Am. 102, 2892-2905 (1997)] was employed to quantitatively predict the experimental data. Effects of varying masker bandwidth, center frequency, modulator bandwidth, modulator type, and signal duration on CMR were examined. In addition, the effect of band limiting the noise before or after modulation was shown to influence the CMR in the same way as a systematic variation of the modulation depth. It is demonstrated that a single-channel analysis, which analyzes only the information from one peripheral channel, quantitatively accounts for the CMR in most cases, indicating that an across-channel process is generally not necessary for simulating results from this class of CMR experiments. True across-channel processes may be found in another class of CMR experiments.

Adult↗
PubMed1999

Continuous assessment of time-varying speech quality.

This paper addresses the question of whether subjects are able to assess the perceived time-varying quality of speech material continuously. A method is introduced which is characterized by a subjective continuous rating of the perceived speech quality by moving a slider along a graphical scale. The usability of this method is illustrated with an experiment in which different sequences of sentences were degraded in quality with a Modulated Noise Reference Unit. The modulation depth was varied with time and the subject's task was to assess the perceived quality. The results indicate that subjects can monitor speech quality variations very accurately with a delay of approximately 1 s. An objective speech quality measure based on an auditory processing model was applied to predict the subjective speech quality results. The speech quality measure qC was modified to allow for time-dependent objective measurement of the speech quality. The averaged subjective response data could be modeled by the scale transformed and low-pass filtered measure qC(t) with a high degree of accuracy.

Adult↗
PubMed1999

Evidence for the distortion product frequency place as a source of distortion product otoacoustic emission (DPOAE) fine structure in humans. I. Fine structure and higher-order DPOAE as a function of the frequency ratio f2/f1.

Critical experiments were performed in order to validate the two-source hypothesis of distortion product otoacoustic emissions (DPOAE) generation. Measurements of the spectral fine structure of DPOAE in response to stimulation with two sinusoids have been performed with normal-hearing subjects. The dependence of fine-structure patterns on the frequency ratio f2/f1 was investigated by changing f1 or f2 only (fixed f2 or fixed f1 paradigm, respectively), and by changing both primaries at a fixed ratio and looking at different order DPOAE. When f2/f1 is varied in the fixed ratio paradigm, the patterns of 2 f1-f2 fine structure vary considerably more if plotted as a function of f2 than as a function of fDP. Different order distortion products located at the same characteristic place on the basilar membrane (BM) show similar patterns for both, the fixed-f2 and fDP paradigms. Fluctuations in DPOAE level up to 20 dB can be observed. In contrast, the results from a fixed-fDP paradigm do not show any fine structure but only an overall dependence of DP level on the frequency ratio, with a maximum for 2f1-f2 at f2/f1 close to 1.2. Similar stimulus configurations used in the experiments have also been used for computer simulations of DPOAE in a nonlinear and active model of the cochlea. Experimental results and model simulations give strong evidence for a two-source model of DPOAE generation: The first source is the initial nonlinear interaction of the primaries close to the f2 place. The second source is caused by coherent reflection from a re-emission site at the characteristic place of the distortion product frequency. The spectral fine structure of DPOAE observed in the ear canal reflects the interaction of both these sources.

Acoustic Stimulation↗
PubMed1999

Evidence for the distortion product frequency place as a source of distortion product otoacoustic emission (DPOAE) fine structure in humans. II. Fine structure for different shapes of cochlear hearing loss.

Distortion product otoacoustic emissions (DPOAE) were recorded from eight human subjects with mild to moderate cochlear hearing loss, using a frequency spacing of 48 primary pairs per octave and at a level L1 = L2 = 60 dBSPL and with a fixed ratio f2/f1. Subjects with different shapes of hearing thresholds were selected. They included subjects with near-normal hearing within only a limited frequency range, subjects with a notch in the audiogram, and subjects with a mild to moderate high-frequency loss. If the primaries were located in a region of normal or near-normal hearing, but DP frequencies were located in a region of raised thresholds, the distortion product 2 f1-f2 was still observable, but the DP fine structure disappeared. If the DP frequencies fell into a region of normal thresholds, fine structure was preserved as long as DPOAE were generated, even in cases of mild hearing loss in the region of the primaries. These experimental results give further strong evidence that, in addition to the initial source in the primary region, there is a second source at the characteristic place of fDP. Simulations in a nonlinear and active computer model for DPOAE generation indicate different generation mechanisms for the two components. The disappearance of DPOAE fine structure might serve as a more sensitive indicator of hearing impairment than the consideration of DP level alone.

Adult↗
PubMed1998

Binaural and monaural auditory filter bandwidths and time constants in probe tone detection experiments.

Auditory filter bandwidths and time constants were obtained with five normal-hearing subjects for different masker configurations both in the frequency and time domain for monaural and binaural listening conditions. Specifically, the masking level in the monaural condition and the interaural correlation in the binaural conditions, respectively, was changed in a sinusoidal, stepwise, and rectangular way in the frequency domain. In the corresponding experiments in the time domain, a sinusoidal and stepwise change of the masker was performed. From these results, a comparison was made across conditions to evaluate the influence of the factors "shape of transition," "monaural versus binaural," "frequency domain versus time domain," and "subject." Also, the respective data from the literature were considered using the same model assumptions and fitting strategy as used for the current data. The results indicate that the monaural auditory filter bandwidths and time constants fitted to the data are consistent across conditions both for the data included in this study and the data from the literature. No consistent relation between individual auditory filter bandwidths and time constants were found across subjects. For the binaural conditions, however, considerable differences were found in estimates of the bandwidths and time constants, respectively, across conditions. The reason for this mismatch seems to be the different detection strategies employed for the various tasks that are affected by the consistency of binaural information across frequency and time. While monaural detection performance appears to be modeled quite well with a linear filter or temporal integration window, this does not hold for the binaural conditions where both larger bandwidth and time constant estimates are found.

Adult↗
PubMed1997

Interaction of otoacoustic emissions with additional tones: suppression or synchronization?

The influence of an external tone on transitory evoked otoacoustic emissions (TEOAE) is investigated. Three different averaging techniques were used with the same acoustic stimulus paradigm. These techniques permitted the separation of those parts of the otoacoustic emission (OAE) that contribute to the transitory evoked otoacoustic emission and those parts of the OAE that are synchronized to the continuous tone. The experiments show that the total energy of the OAE is not reduced in the presence of an additional tone. The 'suppression' of TEOAEs is an effect of synchronization and the subsequent elimination of the 'suppressed' emission in the averaging procedure.

Acoustic Stimulation↗
PubMed1997

Directivity of binaural noise reduction in spatial multiple noise-source arrangements for normal and impaired listeners.

Speech reception thresholds (SRTs) were obtained for different azimuths of up to three interfering sound sources in a simulated ("virtual") spatial arrangement. For the SRT measurements a fast subjective threshold assessment method was used. The reduction in SRT relative to the reference threshold caused by spatial separation of target and interference signal is called the intelligibility level difference (ILD). For normal listeners, the maximum ILD was achieved at 105 degrees or 255 degrees azimuth of a single noise source when the target emanated from the front (0 degree azimuth). The ILD decreased rapidly if the number of interfering signals was increased and if they were located on opposite sides of the head. When using continuous speech as interference, this decrease of ILD was less pronounced. These findings indicate that the binaural system can suppress interference from only one azimuthal region at a time and can utilize temporal pauses in the interference to suppress a second interference from a different direction. For eight sensorineural impaired listeners, the maximum attainable ILD for one interfering noise source was reduced in a way not predictable from the audiogram. The ability to exploit pauses in the signal of an interfering talker was also reduced. Hence, hearing-impaired listeners are handicapped in realistic spatial interference situations both by an increased monaural SRT and a reduced binaural noise reduction ability.

Adult↗
PubMed1997

Relations between notched-noise suppressed TEOAE and the psychoacoustical critical bandwidth.

Narrow-band transitory evoked otoacoustic emissions (TEOAE) were recorded for nine normal hearing subjects in the presence of a broadband tone complex suppressor. Introducing a spectral notch at the frequency of the narrow-band stimulus causes the suppression effect to decrease, the more so the wider the notch. This decrease in suppression permits an estimate of the size of one critical band. One advantage of this approach is that no active participation of the subjects is required. The estimated critical bandwidth is then compared with independent estimates based on a simultaneous masking experiment, using the same stimuli. The two measures of the critical bandwidth coincide well for those six subjects with spontaneous otoacoustic emissions. However, the bandwidth estimate based on the OAE measurements is too large for the other three subjects without spontaneous emissions. Simulations of the suppression effect with a driven van der Pol oscillator with moderate undamping produce critical bandwidth estimates consistent with those observed in the psychoacoustical experiments. This allows an estimate of the "effective" amount of undamping on the basilar membrane that is required to produce the critical bandwidth observable in psychoacoustic experiments.

Acoustic Stimulation↗
PubMed1997

Development and evaluation of a German sentence test for objective and subjective speech intelligibility assessment.

A German sentence test was developed which is comprised of 20 test lists of ten sentences each. The test corpus is a selection from sentences for speech quality evaluation recorded with a male unschooled speaker. Performance-intensity curves were measured for each individual sentence in a speech-simulating babble noise with a total of 40 normal-hearing listeners. Based on these data and the phonemic transcription of the 200 sentences selected from the underlying speech corpus, 20 test lists were composed using a numerical optimization process. These 20 test lists are highly equivalent with respect to their performance-intensity curves, the number of words within each test list, the number of phonemes within each test list, and approximately the frequency distribution of the phonemes which approximates the phoneme frequency distribution of the German language. The equivalence of the respective performance-intensity curves was demonstrated in an independent experiment with 20 normal-hearing listeners. In addition, a comparison was performed between the "objective" intelligibility measurements and two "subjective" speech intelligibility rating methods employing the same materials. As a result, both subjective assessment procedures correlate highly with each other and with the "objective" procedure across sentences. This underlines the applicability and validity of the test in combination with time-saving subjective assessment methods. Moreover, the variability in performance across different sentences correlates inversely with the RMS level of the respective sentence. This indicates that an adjustment of sentence material with respect to RMS level already yield reasonably homogeneous test material with respect to intelligibility.

Adult↗
PubMed1997

Modeling auditory processing of amplitude modulation. I. Detection and masking with narrow-band carriers.

This paper presents a quantitative model for describing data from modulation-detection and modulation-masking experiments, which extends the model of the "effective" signal processing of the auditory system described in Dau et al. [J. Acoust. Soc. Am. 99, 3615-3622 (1996)]. The new element in the present model is a modulation filterbank, which exhibits two domains with different scaling. In the range 0-10 Hz, the modulation filters have a constant bandwidth of 5 Hz. Between 10 Hz and 1000 Hz a logarithmic scaling with a constant Q value of 2 was assumed. To preclude spectral effects in temporal processing, measurements and corresponding simulations were performed with stochastic narrow-band noise carriers at a high center frequency (5 kHz). For conditions in which the modulation rate (fmod) was smaller than half the bandwidth of the carrier (delta f), the model accounts for the low-pass characteristic in the threshold functions [e.g., Viemeister, J. Acoust. Soc. Am. 66, 1364-1380 (1979)]. In conditions with fmod > delta f/2, the model can account for the high-pass characteristic in the threshold function. In a further experiment, a classical masking paradigm for investigating frequency selectivity was adopted and translated to the modulation-frequency domain. Masked thresholds for sinusoidal test modulation in the presence of a competing modulation masker were measured and simulated as a function of the test modulation rate. In all cases, the model describes the experimental data to within a few dB. It is proposed that the typical low-pass characteristic of the temporal modulation transfer function observed with wide-band noise carriers is not due to "sluggishness" in the auditory system, but can instead be understood in terms of the interaction between modulation filters and the inherent fluctuations in the carrier.

Acoustics↗
PubMed1997

Modeling auditory processing of amplitude modulation. II. Spectral and temporal integration.

A multi-channel model, describing the effects of spectral and temporal integration in amplitude-modulation detection for a stochastic noise carrier, is proposed and validated. The model is based on the modulation filterbank concept which was established in the accompanying paper [Dau et al., J. Acoust. Soc. Am. 102, 2892-2905 (1997)] for modulation perception in narrow-band conditions (single-channel model). To integrate information across frequency, the detection process of the model linearly combines the channel outputs. To integrate information across time, a kind of "multiple-look" strategy, is realized within the detection stage of the model. Both data from the literature and new data are used to validate the model. The model predictions agree with the results of Eddins [J. Acoust. Soc. Am. 93, 470-479 (1993)] that the "time constants" associated with the temporal modulation transfer functions (TMTF) derived for narrow-band stimuli do not vary with carrier frequency region and that they decrease monotonically with increasing stimulus bandwidth. The model is able to predict masking patterns in the modulation-frequency domain, as observed experimentally by Houtgast [J. Acoust. Soc. Am. 85, 1676-1680 (1989)]. The model also accounts for the finding by Sheft and Yost [J. Acoust. Soc. Am. 88, 796-805 (1990)] that the long "effective" integration time constants derived from the data are two orders of magnitude larger than the time constants derived from the cutoff frequency of the TMTF. Finally, the temporal-summation properties of the model allow the prediction of data in a specific temporal paradigm used earlier by Viemeister and Wakefield [J. Acoust. Soc. Am. 90, 858-865 (1991)]. The combination of the modulation filterbank concept and the optimal decision algorithm proposed here appears to present a powerful strategy for describing modulation-detection phenomena in narrow-band and broadband conditions.

Auditory Perception↗
PubMed1996

Speech intelligibility prediction in hearing-impaired listeners based on a psychoacoustically motivated perception model.

Sensorneural hearing-impaired listeners suffer severely from deterioration in the processing and internal representation of acoustic signals. In order to understand this deterioration in detail, a numerical perception model was developed which is based on current functional models of the signal processing in the auditory system. To test this model, the individual's speech intelligibility in quiet and in noise was predicted. The primary input parameter of the model is the precisely measured audiogram of each listener. In a refined version of the model, additional input parameters are derived from predicting the individual's temporal forward masking and notched-noise measurements with the same model assumptions. The predictions of the perception model were compared with those of the articulation index (AI) and the speech transmission index (STI). The accuracy of prediction with the perception model is in the same range as with the AI and the STI. The model does not require a calibration function and has the advantage of a greater flexibility in including different processing deficits associated with hearing impairment. However, it requires more time for computation. For the hearing-impaired listeners examined so far the individually measured psychoacoustical parameters have only a secondary effect on the prediction as compared to the audiogram. Nevertheless, the underlying model is a first step toward a quantitative understanding of speech intelligibility and helps to distinguish between the influence of the "attenuation" and the "distortion" component of the hearing loss.

Adult↗