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Biomedical subjects

M Kompis

Publications and source records attributed to M Kompis.

6 recordsLinked to original sources
PubMed2000

A novel real-time noise reduction system for the assessment of evoked otoacoustic emissions.

A novel noise reduction method and apparatus to be used in connection with the measurement of evoked otoacoustic emissions (EOAE) are presented. The noise reduction method is based on an adaptive noise canceller and requires a noise-only reference microphone placed in the vicinity of the OAE-probe. The method was implemented in real time on a custom built digital signal processing system using an Analog Devices ADSP-2181 digital signal processor. The system interfaces seamlessly with a commercial EOAE acquisition system. Results of a series of experiments show than noise reductions of 7-8 dB can be reached.

Acoustic Stimulation↗
PubMed1999

Performance of compressed analogue (CA) and continuous interleaved sampling (CIS) coding strategies for cochlear implants in quiet and noise.

Speech understanding with compressed analogue (CA) and continuous interleaved sampling (CIS) coding strategies for cochlear implants was compared in quiet and in noise at signal-to-noise ratios (SNRs) of 15, 10 and 5 dB. The speech recognition of three experienced users of the Ineraid cochlear implant (CA coding strategy) was assessed using a set of sentence, vowel and consonant tests. Three weeks after the fitting of a CIS processor, the tests were repeated with the new device. Speech recognition scores for the sentence and consonant tests tended to be higher with the CIS processor in no or little noise, but lower in the test situations with less favourable SNRs, when compared to the CA processor (average score differences for the consonant test: +7.8% correct at 15 dB SNR; -6.8% correct at 5 dB SNR; p = 0.05). Results for the vowel test were slightly lower on average for the CIS processing strategy at all SNRs. A possible explanation for the differences in performance between CIS and CA in the consonant and sentence tests at different SNRs is the generally higher free-field threshold associated with the CA coding strategy, which may act as a single-channel noise suppression.

Adolescent↗
PubMed1998

Effects of breathing pathways on tracheal sound spectral features.

The spectra of sounds recorded over the trachea of adults typically reveal peaks near 700 and 1500 Hz. We assessed the anatomical determinants of these peaks and the conditions contributing to their presence. We studied five adult subjects with normal lung function, measuring sounds at the suprasternal notch and on the right cheek. The subjects breathed at target airflows of 15 and at 30 ml sec(-1) kg(-1) both through the mouth with nose clips and then through the mouth and nose using a cushioned face mask. The mouth breathing maneuvers were performed with three lengths (3.6, 21.1 and 38.6 cm) of 2.6 cm diameter tubing between the mouth and the pneumotachograph. The nose breathing maneuver was performed with the longest tube (between the mask and pneumotachograph). The signals occurring at the target flows +/- 20% were used to create averaged, spectral estimates. We found that all subjects had two predominant spectral peaks; a approximately 700 Hz peak loudest over the cheek and a approximately 1500 Hz peak loudest over the trachea. The frequency of both peaks negatively correlated with body height (and presumably, airway length). There was no systematic effect of breathing phase, flow rate or length of the tube connecting the mouth to the pneumotachograph on the spectral peaks. Breathing into the mask and breathing through the nose did markedly alter the spectra. We conclude that the higher tracheal sound peak reflects resonance within the major airways and is relatively independent of extrathoracic influences during mouth breathing through a tube.

Adult↗
PubMed1997

Computer-based lung sound simulation.

An algorithm for the simulation of normal and pathological lung sounds is developed. The simulation algorithm is implemented on a personal computer as well as on a digital signal processor system in real time. Normal, bronchial and tracheal breathing sounds can be generated, and continuous and discontinuous adventitious lung sounds can be added. The attributes of the individual sound components, such as loudness, frequency, duration or number of occurrences within one breathing cycle, are controlled by the user. The quality of the simulations is evaluated by sending audio tapes to 15 experienced pulmonary physicians for a formal assessment. Each tape contains five simulated lung sounds and five real lung sounds from a commercially available teaching tape, presented in random order. Simulated lung sounds are slightly better rated in terms of realism and signal quality when compared to the recordings from the teaching tape. The differences are, however, not significant. 13 out of the 15 physicians feel that computer-based lung sound simulators would be a useful and desirable teaching tool for auscultation courses.

Algorithms↗
PubMed1993

Digital signal processing (DSP) applications for multiband loudness correction digital hearing aids and cochlear implants.

Single-chip digital signal processors (DSPs) allow the flexible implementation of a large variety of speech analysis, synthesis, and processing algorithms for the hearing impaired. A series of experiments was carried out to optimize parameters of the adaptive beamformer noise reduction algorithm and to evaluate its performance in realistic environments with normal-hearing and hearing-impaired subjects. An experimental DSP system has been used to implement a multiband loudness correction (MLC) algorithm for a digital hearing aid. Speech tests in quiet and noise with 13 users of conventional hearing aids demonstrated significant improvements in discrimination scores with the MLC algorithm. Various speech coding strategies for cochlear implants were implemented in real time on a DSP laboratory speech processor. Improved speech discrimination performance was achieved with high-rate stimulation. Hybrid strategies incorporating speech feature detectors and complex decision algorithms are currently being investigated.

Adult↗