PubMed Health⌕ Search

Biomedical subjects

B Kollmeier

Publications and source records attributed to B Kollmeier.

32 records · Page 2Linked to original sources

The effect of multichannel dynamic compression on speech intelligibility.

In the recent literature on dynamic compression in hearing aids, a controversial discussion about the benefit of multichannel compression with short time constants has been raised. In this paper, some experimental results are contributed to the discussion: For low signal-to-noise ratios, a detrimental effect of fast multichannel compression on speech intelligibility in normal hearing listeners is found. However, a distinct degradation in speech quality and virtually no decrease in speech intelligibility is observed for comparatively high signal-to-noise ratios and compression ratios of up to 3. Since the predictions of the Speech Transmission Index deviate substantially from these results, its applications as an objective measure of intelligibility may not be extended to compressed speech.

Hearing Aids↗

Narrowband stimulation and synchronization of otoacoustic emissions.

Tone-burst-evoked otoacoustic emissions were obtained for several normal listeners with and without spontaneous otoacoustic emission using an optimized tone burst. The dependence of the response amplitude on stimulation level shows a linear increase below a certain value close to the threshold in quiet and levels off at higher levels exhibiting only small differences between non-linear and linear averaging modus. In addition, the latency of the response tends to decrease and the sharpness of the resonance decreases with increasing stimulation level. Synchronization tuning-curves were obtained using a fixed tone burst to elicit an evoked otoacoustic emission and a probe tone at different frequencies for synchronizing the SOAE. These synchronization tuning curves exhibit relatively sharp resonance characteristics (Q3 varying between 3 and 8) for subjects with spontaneous otoacoustic emissions and less sharp tuning (Q3 varying between 1 and 3) for subjects without spontaneous otoacoustic emissions. These experimental results could be reproduced very well with a model of a single non-linear Van-der-Pol-oscillator with the appropriate parameters. The results indicate that spontaneous otoacoustic emissions and narrow-band-evoked otoacoustic emissions are generated by the same mechanism which can be modelled as a self-sustained oscillator.

Acoustic Stimulation↗

Chirp evoked otoacoustic emissions.

The principles of short frequency sweeps (chirps) and their application to evoke transiently evoked otoacoustic emissions (TEOAE) are developed in comparison to using standard click stimuli. In contrast to click stimuli, chirp signals have the advantage of stimulating a freely selectable frequency range. In addition, chirp signals contain more energy than a click stimulus with the same maximum amplitude. The effects of different stimuli on TEOAE were investigated in normal hearing and hearing-impaired subjects. Using wide-band chirp signals yields a better signal-to-noise ratio compared to click stimulation. In addition, the stimulation of selected regions of the basilar membrane with frequency-limited chirps evokes TEOAE with frequency components that lie within the stimulated frequency range. The characteristic fine structure of this spectrum was found to be independent of the stimulus applied. The utilization of chirp stimuli appears to be useful for evoking TEOAE in, e.g., clinical applications.

Acoustic Stimulation↗

Speech enhancement based on physiological and psychoacoustical models of modulation perception and binaural interaction.

A novel approach for analyzing and filtering speech is described and evaluated which utilizes the "modulation spectrogram," i.e., the two-dimensional representation of modulation frequencies versus center frequency as a function of time. This approach is based on physiological findings of a tonotopical organization of modulation frequencies perpendicular to carrier frequencies as well as psychoacoustical findings of "modulation tuning curves." In addition, an interaction is assumed between the representation of modulation frequencies and the representation of auditory space as described by physiological and psychological models of binaural hearing. A noise-reduction algorithm based on this approach was implemented and tested which enhances or suppresses each combination of modulation frequency and center frequency according to its phase and intensity relation between the two input signals (i.e., both stereo channels of a dummy-head recording). When tested in several situations with interfering speakers and background noise both in anechoic and reverberant environment, the algorithm provided a small but a very robust increase in speech intelligibility which corresponds to approximately 2 dB in signal-to-noise ratio. Possible applications of this algorithm are noise reduction in adverse acoustical situations, digital hearing aids, processing schemes and preprocessing for speech recognition.

Adult↗

Binaural noise-reduction hearing aid scheme with real-time processing in the frequency domain.

A binaural noise-reduction algorithm for suppressing lateral noise sources was implemented in real-time and tested with normal and hearing-impaired subjects. With this algorithm, frequency bands with interaural time and amplitude differences characteristic for the "target" direction of the desired speech sound reach both ears without any attenuation. However, frequency bands with an interaural time and amplitude difference deviating from these desired values are attenuated. Speech quality assessments as well as speech intelligibility tests demonstrate a significant noise reduction of up to about 5 dB in signal-to-noise ratio for a dummy-head recorded "cocktail-party situation" in an anechoic chamber. For two additional acoustical situations, however, the subjective speech quality from the algorithm degrades with increasing reverberation, while the output from a real-time "dereverberation" algorithm was ranked higher with increasing reverberation. The results with hearing-impaired listeners suggest a combination of both algorithms, while stressing the importance of binaural listening and binaural hearing aids for the rehabilitation of impaired listeners in noisy environments.

Acoustic Stimulation↗

Real-time multiband dynamic compression and noise reduction for binaural hearing aids.

A multi-signal-processor set-up is introduced that is used for real-time implementation of digital hearing aid algorithms that operate on stereophonic (i.e., binaural) input signals and perform signal processing in the frequency domain. A multiband dynamic compression algorithm was implemented which operates in 24 critical band filter channels, allows for interaction between frequency bands and stereo channels, and is fitted to the hearing of the individual patient by a loudness scaling method. In addition, a binaural noise reduction algorithm was implemented that amplifies sound emanating from the front and suppresses lateral noise sources as well as reverberation. These algorithms were optimized with respect to their processing parameters and by minimizing the processing artifacts. Different versions of the algorithms were tested in six listeners with sensorineural hearing impairment using both subjective quality assessment methods and speech intelligibility measurements in different acoustical situations. For most subjects, linear frequency shaping was subjectively assessed to be negative, although it improved speech intelligibility in noise. Additional compression was assessed to be positive and did not deteriorate speech intelligibility as long as the processing parameters were fitted carefully. All noise reduction strategies employed here were subjectively assessed to be positive. Although the suppression of reverberation only slightly improved speech intelligibility, a combination of directional filtering and dereverberation provided a substantial improvement in speech intelligibility for most subjects and for a certain range of signal-to-noise ratios. The real-time implementation was very helpful in optimizing and testing the algorithms, and the overall results indicate that carefully designed and fitted binaural hearing aids might be very beneficial for a large number of patients.

Adult↗

[Evoked otoacoustic emissions as a screening test for hearing evaluation in newborn and premature infants?].

The purpose of this study was the evaluation of evoked otoacoustic emissions (EOAE) for the screening of infant hearing. EOAE and brainstem-evoked response audiometry (BERA) were performed in 40 infants at a special care unit at the Universitäts-Klinik, Göttingen. Both examinations were performed under non-optimal conditions, reflecting a realistic screening situation. The purpose was to determine robust analysis criteria and possible error sources. A comparison of both tests shows that sensitivity and specificity are insufficient for reliable screening. The EOAE test does not yield the threshold of hearing. There is no sharp boundary with the BERA threshold for the detectability of an EOAE. Sensitivity and specificity have not been accurately determined so far due to the relatively small number of infants with impaired hearing tested. Nevertheless, the results show that EOAE should be applied as a supplement to conventional audiometry for infants. It is capable of excluding conductive and severe cochlear hearing losses and is by itself not a reliable screening test.

Audiometry, Evoked Response↗

Auditory filter bandwidths in binaural and monaural listening conditions.

The shape and the effective bandwidth of the auditory filter at 500 Hz was examined for binaural and monaural tone-in-noise detection experiments in four normal listeners. In the binaural condition, a broadband noise with an interaural phase difference of 0 below and an interaural phase difference of pi above a certain "edge frequency" was employed to mask a 500-Hz probe tone with an interaural phase pi (denoted as No pi S pi). The threshold of the probe tone as a function of the edge frequency in this configuration and in a configuration with an inverted interaural phase of the masker (denoted as N pi oS pi) was fitted by assuming different filter shapes and optimizing their respective parameters. In an analogous monaural experiment, the spectral power density of the masker was 15 dB lower below the "edge frequency" or 15 dB lower above this frequency, respectively. Several filter characteristics with two free parameters describe the data almost equally well. Their equivalent rectangular bandwidths (ERB) show considerably more variations between filter shapes than the 10-dB bandwidth and the 90% bandwidth values (i.e., the bandwidths encompassing 90% of the integrated area above and below the center frequency). This indicates that either of these two bandwidth parameters is more appropriate for comparing auditory filter bandwidths than the ERB. For the rounded exponential filter, the 90% bandwidth averages to 147 Hz in the binaural and to 125 Hz in the monaural condition. These values are up to 12% higher if off-frequency detection is accounted for. Our general finding of auditory filter bandwidths in the binaural conditions exceeding the monaural bandwidths by approximately 20% may be caused by two factors: First, off-frequency detection may be performed in monaural, but not in binaural detections tasks and second, the random interaural mismatch in binaural noise reduction processes fluctuates slowly and thus modulates and spectrally smears the output signal of the binaural noise reduction process.

Adult↗

[Evoked otoacoustic emissions in adults. Criteria for evaluation in clinical use].

The work here presents the first part of a prospective study regarding the clinical use of evoked otoacoustic emissions (EOAE) in adults. Sixty subjects with normal hearing and 160 patients suffering from cochlear hearing loss were tested. The results were used to develop and optimize analysis criteria for the emissions, based on their physical properties. A short-time Fourier analysis was performed so that the EOAE intensity in time and frequency domains could be observed simultaneously. A comparison of this data with the individual thresholds of hearing showed the importance of the EOAE level. However, the bandwidth of the EOAE demonstrated an even steeper transition between normal and hearing-impaired subjects. These findings suggest that this bandwidth is a better criterion for the detection of an EOAE. On the other hand, both parameters correlated weakly with the hearing threshold and the differences between subjects were very large. A prediction of hearing loss based on EOAE results is impossible with the wideband click stimulation used here.

Acoustic Stimulation↗

Binaural forward and backward masking: evidence for sluggishness in binaural detection.

The threshold of a short interaurally phase-inverted probe tone (20 ms, 500 Hz, S pi) was obtained in the presence of a 750-ms noise masker that was switched after 375 ms from interaurally phase-inverted (N pi) to interaurally in-phase (No). As the delay between probe-tone offset and noise phase transition is increased, the threshold decays from the N pi S pi threshold (masking level difference = 0 dB) to the No S pi threshold (masking level difference = 15 dB). The decay in this "binaural" situation is substantially slower than in a comparable "monaural" situation, where the interaural phase of the masker is held constant (N pi), but the level of the masker is reduced by 15 dB. The prolonged decay provides evidence for additional binaural sluggishness associated with "binaural forward masking." In a second experiment, "binaural backward masking" is studied by time reversing the maskers described above. Again, the situation where the phase is switched from No to N pi exhibits a slower transition than the situation with constant interaural phase (N pi) and a 15-dB increase in the level of the masker. The data for the binaural situations are compatible with the results of a related experiment, previously reported by Grantham and Wightman [J. Acoust. Soc. Am. 65, 1509-1517 (1979)] and are well fit by a model that incorporates a double-sided exponential temporal integration window.

Adult↗

Speech enhancement by filtering in the loudness domain.

Several attempts that employed dynamic compression to compensate for cochlear impairment in combination with recruitment have failed to improve speech intelligibility. One reason is the attenuation of dynamic intensity modulations in the range of 1 to 50 Hz which contribute significantly to speech discrimination. In the approach presented here, these modulations were enhanced rather than attenuated by an envelope highpass filter operating in the loudness domain as follows: the output of a filterbank was lowpass filtered to extract the envelope, converted to loudness in accordance with normal hearing, highpass-filtered and converted back to the 'desired' envelope by the loudness characteristic of each individual hearing-impaired subject. The output signal was finally obtained by summing the outputs of the filterbank channels after correcting their amplitudes. The algorithm was implemented by a computer to simulate a digital hearing aid fitted to the loudness-scaling function of each individual. Loudness scaling and speech-discrimination data in quiet and in noisy conditions are presented for 2 impaired subjects with and without highpass filtering in the loudness domain. The results indicate that loudness compensation without highpass filtering improves speech intelligibility for a wide range of input levels by achieving an optimum gain characteristic at all input levels. The highpass filter applied to the envelope has no noticeable effect on speech discrimination either in quiet or in noise.

Algorithms↗

Speech intelligibility enhancement by interaural magnification.

Hearing-impaired persons suffer from loss of speech intelligibility in noisy environments as well as from a decrease in binaural performance (e.g., interaural time and intensity discrimination). In order to improve speech intelligibility for spatially separated signal and noise sources by compensating for the reduced binaural performance, we implemented an algorithm increasing binaural dissimilarity on a digital signal processor. This algorithm is a modification of the proposal by Durlach & Pang (J. Acoust. Soc. Am. 1986; 80: 1849-1850). Speech discrimination experiments were performed under simulated spatial conditions with recordings of the speech material and the interfering noise source using a dummy head. With increasing complexity of the spatial masking condition, the binaural advantage as well as the benefits from the algorithm decreased for normal listeners. For four sensorineurally hearing-impaired listeners, only one spatial condition was tested. In addition, a battery of binaural tests (binaural masking level difference, interaural time and intensity discrimination) was performed. The results indicate that the benefits from the signal-processing technique are correlated with the remaining binaural performance.

Adult↗

Adaptive staircase techniques in psychoacoustics: a comparison of human data and a mathematical model.

Data from a simple tone-in-noise simultaneous masking task were used to evaluate each of two common adaptive staircase rules (a "1 up 2 down" rule and a "1 up 3 down" rule) and the parameter estimation by sequential testing (PEST) technique in combination with each of two psychophysical procedures [a two-alternative forced-choice (2AFC) and a three-alternative forced-choice (3AFC) procedure]. These human data were compared to predictions generated by a mathematical model based on Markov theory. The model predicts that threshold estimates obtained with the adaptive techniques should be equal to those derived with equivalent "fixed signal level" techniques. However, the human data indicate that the adaptive techniques tend to yield lower thresholds. The model predicts that the standard error of a threshold estimate obtained from an adaptive technique will decrease and approach zero as the number of trials used to compute the estimate increases. The human data show greater variability than predicted and approach a nonzero value as the number of trials increases. The predictions of the model suggest that the commonly used combination of the 2AFC procedure and the 1 up 2 down rule is the least efficient method of estimating a threshold and that the 3AFC procedure in combination with the 1 up 3 down rule is the most efficient method. The human data are less consistent, but generally show the combination of the 2AFC procedure and the 1 up 2 down rule to be one of the least efficient methods. Possible explanations for the differences between the model's predictions and the human data, as well as suggestions for laboratory practice, are discussed.

Adult↗

Detection of the acoustic reflex below 80 dB HL.

A new method for detecting the acoustic reflex that utilizes standard otoacoustic emissions recording techniques is introduced and discussed. Two successive identical tone bursts of 100 ms duration and 10 ms interstimulus interval are presented in the occluded ear canal at a repetition rate of one per second. If the acoustic reflex is elicited, the contraction of the stapedius muscle is delayed with respect to the onset of the first stimulus. Hence, the acoustic compliance in the ear canal decreases primarily during the second stimulus. The difference of the microphone signals produced by the two stimuli is computed and averaged across a certain number of repetitions of the sequence. The presentation level is increased until this difference is larger than -40 dB (with respect to the stimulus level) and if its signal-to-noise ratio exceeds 20 dB. For normal-hearing subjects, the acoustic reflex threshold measured with this method is on average 8 dB lower than in a standard clinical setup. In 5 out of the 10 tested hearing-impaired subjects, the new method could detect an acoustic reflex at one or more frequencies where no reflex was detected in the clinical setup.

Adult↗