The paper deals with relationship between speech recognition and objective parameters of enclosures. Six enclosures were chosen: a church, an assembly hall of a music school, two courtrooms of different volumes, a typical auditorium and a university concert hall. Dirac 4.1 software was used to record impulse responses (IRs) in the chosen measurement points of each enclosure. On this base, the following objective parameters of the enclosure were determined: Reverberation Time (RT), Early Decay Time (EDT), Weighted Clarity (C50) and Speech Transmission Index (STI). A convolution of the IRs with logatome tests and the Polish Sentence Test (PST) was made. Logatome recognition and speech reception threshold (SRT – i.e., SNR yielding 50% speech recognition) were evaluated and their dependence on the objective parameters were determined. Generally a linear relationship between logatome or SRT and RT or EDT was found. However, speech recognition was nonlinearly related (according to psychometric function) to STI values. The most sensitive range of the logatome and sentence recognition relative to STI changes corresponded to the middle range of STI values. Below and above this range, logatome and sentence recognition were much less dependent of STI changes.
The aim of this work was to measure subjective speech intelligibility in an enclosure with a long reverberation time and comparison of these results with objective parameters. Impulse Responses (IRs) were first determined with a dummy head in different measurement points of the enclosure. The following objective parameters were calculated with Dirac 4.1 software: Reverberation Time (RT), Early Decay Time (EDT), weighted Clarity (C50) and Speech Transmission Index (STI). For the chosen measurement points, a convolution of the IRs with the Polish Sentence Test (PST) and logatome tests was made. PST was presented at a background of a babble noise and speech reception threshold – SRT (i.e. SNR yielding 50% speech intelligibility) for those points were evaluated. A relationship of the sentence and logatome recognition vs. STI was determined. It was found that the final SRT data are well correlated with speech transmission index (STI), and can be expressed by a psychometric function. The difference between SRT determined in condition without reverberation and in reverberation conditions appeared to be a good measure of the effect of reverberation on speech intelligibility in a room. In addition, speech intelligibility, with and without use of the sound amplification system installed in the enclosure, was compared.
The paper presents the results of sentence and logatome speech intelligibility measured in rooms with induction loop for hearing aid users. Two rooms with different acoustic parameters were chosen. Twenty two subjects with mild, moderate and severe hearing impairment using hearing aids took part in the experiment. The intelligibility tests composed of sentences or logatomes were presented to the subjects at fixed measurement points of an enclosure. It was shown that a sentence test is more useful tool for speech intelligibility measurements in a room than logatome test. It was also shown that induction loop is very efficient system at improving speech intelligibility. Additionally, the questionnaire data showed that induction loop, apart from improving speech intelligibility, increased a subject’s general satisfaction with speech perception.
The main purpose of this investigation was to measure the effect of contralateral acoustic stimulation (CAS) on distortion product otoacoustic emission (DPOAE) in twenty human ears, for a ratio of primary tones f2/f1 = 1.22 and a wide frequency range of f2 (1.4–9 kHz), for two intensity levels of primary tones (L1 = 60 dB SPL; L2 = 50 dB SPL and L1 = 70 dB SPL; L2 = 60 dB SPL) and two intensity levels of CAS (50 and 60 dB SPL). It was found that in the presence of CAS, in the majority of cases the DPOAE level decreased (suppression), but it might also increase (enhancement) or remain unchanged depending on the frequency. The mean suppression level of the component of the frequency fDP = 2f1 − f2 might be approximated by a linearly decreasing function of the f2 frequency of primary tones. The slope of this function was negative and increased with an increase of the contralateral stimulation level. The higher was the contralateral noise level the greater was the suppression. For the fDP level below about 15 dB SPL, suppression was observed in a substantial number of measurement cases (in about 85% of all measured cases on average). When the fDP level was higher than 15 dB SPL, only suppression (not enhancement) was observed.
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The role of spectral enhancement in identification of Polish vowels is investigated in this study. Identification scores of natural (unprocessed) vowels (Experiment 1) and spectrally enhanced vowels (Experiment 2) were measured for normal-hearing subjects (NHS) and for subjects with hearing impairment of cochlear origin (HIS). The NHS group demonstrated almost 100% vowel-identification scores for natural vowels. Instead, for the HIS group, an overall average identification scores were 85% for natural vowels, 96% for spectrally enhanced vowels. It has been shown that the spectral enhancement applied has improved the average identification score for subjects with sensorineural hearing losses. However, the degree of improvement clearly depended on the subjects, the type of vowel, and the talker.
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The study reported was undertaken as an attempt to improve the intelligibility of selected speech signals (numerical test) masked by a speech-shaped noise, by a proposed algorithm of the speech signal processing based on the spectral contrast enhancement and binaural procedure. The spectral contrast enhancement involved the transformation of the speech signal spectrum to achieve an increase in the level of the amplitude between subsequent minima and maxima of individual formants. The binaural procedure involved the determination of the interaural time difference (ITD) and the interaural intensity difference (IID) in order to select and enhance the fragments reaching the listener from the front. The intelligibility of speech processed by the algorithm was compared to that of the unprocessed speech for different spatial separations between the signal and the noise sources and for a few signal to noise ratios. The results have shown that the intelligibility of the speech presented against the noise significantly depends on the angle between the directions of the signal and the noise. The intelligibility of speech is much larger when the noise and the speech reach the observer from different directions, i.e. when their sources are spatially separated. In general, the algorithm improves the intelligibility of numerical tests presented at the background of the noise by a few percent and the greatest improvement has been observed for low signal to noise ratios. The algorithm performance was different for each hearing impaired subject with binaural hearing loss. To get an objective assessment of the algorithm performance it should be optimised and tested on a larger group of subjects for more diverse sound material.
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The study is devoted to determination of the shape of the auditory filters in subjects with sensorineural hearing loss. Apart from the classical sensorineural hearing loss, changes in the auditory filter shapes have been analysed in the subject diagnosed with dead regions. The dead region is an area on the basilar membrane over which the functioning of the inner hair cells and/or neurones innervating them has ceased. This type of hearing impairment means that the information on the sounds whose frequencies correspond to the dead region of the basilar membrane are to a very limited degree if at all, transmitted to higher levels of the auditory path. This transmission, if happens, is performed through the auditory filters at the centre frequency other than that of the signal. This phenomenon and the fact that in the dead region the hearing loss is theoretically infinite, affect the transmittance of the auditory filters. Results of the study reported here have shown that in general, the subjects with sensorineural hearing loss develop broadening of the auditory filter accompanied by reducing of its dynamics. This fact explains a considerable decrease in speech intelligibility presented at a background of a noise. In the subject with the dead regions the broadening of the filters was the greatest in the region of the dead one. The results also indicate the lack of correlation between the shape and width of the auditory filter and the shape of the audiogram.
The main objective of this study was to determine the binaural interaction in modulation frequency domain for FM signals. The results have show that for interaural differences in modulation frequency, except for fast frequency deviation, the subjects perceived additionally slow frequency deviation with rate corresponding to interaural modulation frequency. The range of this deviation depended on interaural differences in modulation rate.
The investigations focused on the binaural perception of amplitude modulated (AM) and frequency modulated (FM) signals. They are comprised of two experiments. In the first experiment binaurally perceived (matched) modulation depth for AM signals was determined under diotic conditions (i.e. for the same values of modulation depth coefficient, m, presented to the left (ml) and right (mr) ears) and under dichotic conditions (i.e. for different values of these coefficients ml [różne od] mr). The measurements were made for the interaural differences in modulation depth coefficient [delta]m, changing from 0 to 100% and a few selected modulating frequencies (4, 64 and 128 Hz) and carrier frequencies (250 and 1000 Hz). In the second experiment binaurally perceived (matched) frequency deviation of FM signals was determined under diotic conditions (i.e. for the same values of frequency deviation, [delta]f, presented to the left (delta f_l) and right (delta fr) ear (delta fl =delta fr) and under dichotic conditions (i.e. for different values of this deviation (delta fl delta delta fr)). The measurements were made for the interaural differences of frequency deviation changing from 0 to 20 Hz; a few selected modulating frequencies (32, 64 and 128 Hz) and carrier frequencies (500 and 1000 Hz). It was found in Experiment I that for small interaural differences in modulation depth, delta m, the binaurally perceived modulation depth, m, is equal to the arithmetic mean of the depths presented to the left and right ears, whereas for large values of delta m, the value of m is smaller than the mean. The results of Experiment II revealed that the binaurally perceived frequency deviation is a linear function of interaural differences of this deviation and is equal to the arithmetic mean of deviations presented to the left and right ears.
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