Michael A. Stone

Age-group differences in speech identification despite matched audiometrically normal hearing: contributions from auditory temporal processing and cognition

Hearing loss with increasing age adversely affects the ability to understand speech, an effect that results partly from reduced audibility. The aims of this study were to establish whether aging reduces speech intelligibility for listeners with normal audiograms, and, if so, to assess the relative contributions of auditory temporal and cognitive processing. Twenty-one older normal-hearing (ONH; 60–79 years) participants with bilateral audiometric thresholds ≤ 20 dB HL at 0.125–6 kHz were matched to nine young (YNH; 18–27 years) participants in terms of mean audiograms, years of education, and performance IQ. Measures included: (1) identification of consonants in quiet and in noise that was unmodulated or modulated at 5 or 80 Hz; (2) identification of sentences in quiet and in co-located or spatially separated two-talker babble; (3) detection of modulation of the temporal envelope (TE) at frequencies 5–180 Hz; (4) monaural and binaural sensitivity to temporal fine structure (TFS); (5) various cognitive tests. Speech identification was worse for ONH than YNH participants in all types of background. This deficit was not reflected in self-ratings of hearing ability. Modulation masking release (the improvement in speech identification obtained by amplitude modulating a noise background) and spatial masking release (the benefit obtained from spatially separating masker and target speech) were not affected by age. Sensitivity to TE and TFS was lower for ONH than YNH participants, and was correlated positively with speech-in-noise (SiN) identification. Many cognitive abilities were lower for ONH than YNH participants, and generally were correlated positively with SiN identification scores. The best predictors of the intelligibility of SiN were composite measures of cognition and TFS sensitivity. These results suggest that declines in speech perception in older persons are partly caused by cognitive and perceptual changes separate from age-related changes in audiometric sensitivity.

New version of the TEN test with calibrations in dB HL.

Objective: To develop a new version of the threshold-equalizing-noise (TEN) test for the diagnosis of dead regions, with levels calibrated in dB HL rather than dB SPL, and with levels corresponding to the dial readings on the audiometer. Design: The spectral shape of the noise required to give equal masked thresholds at all frequencies, when expressed in dB HL, was derived by two calculation methods and by empirical measurements of the electrical output of audiometers using TDH50 earphones and TDH39 earphones. To reduce the loudness of the noise and to minimize distortion generated in the audiometer and/or earphone, the noise was bandlimited between 354 and 6500 Hz. In addition, the noise was synthesized using a method that leads to a low crest factor (ratio of peak to root-mean-square value). This further reduced audiometer/earphone distortion and allowed higher levels per ERBN; ERBN is the equivalent rectangular bandwidth of the auditory filter at 1 kHz, as determined in young, normally hearing subjects. The test tone frequencies were limited to the range 500 to 4000 Hz. Subjects with normal or near-normal hearing were tested by using a noise level of 60 dB HL/ERBN to assess whether the noise did lead to equal masked thresholds in dB HL for all audiometric frequencies from 500 to 4000 Hz. Thresholds in the TEN were measured by means of manual audiometry with a 2 dB final step size. Results: The mean masked thresholds were almost constant across frequency when expressed in dB HL and were within 0.5 dB of the noise level per ERBN. For a single noise level, the test takes approximately 5 minutes per ear to administer. Conclusions: The new TEN test has the following advantages over the original version (which used levels calibrated in dB SPL): (1) All levels are expressed in dB HL. Thus, absolute thresholds only need to be measured once. (2) Calibration is such that both the noise level/ERBN and the test tone levels correspond to the values indicated on the audiometer. This makes the test simpler to apply and reduces the likelihood of errors. (3) The noise bandwidth is restricted, and the noise has a low crest factor. This allows the noise level/ERBN to be increased while avoiding distortion, excessive loudness, and possible further damage to hearing.

Benefits of linear amplification and multichannel compression for speech comprehension in backgrounds with spectral and temporal dips

People with cochlear hearing loss have markedly higher speech-receptions thresholds (SRTs) than normal for speech presented in background sounds with spectral and/or temporal dips. This article examines the extent to which SRTs can be improved by linear amplification with appropriate frequency-response shaping, and by fast-acting wide-dynamic-range compression amplification with one, two, four, or eight channels. Eighteen elderly subjects with moderate to severe hearing loss were tested. SRTs for sentences were measured for four background sounds, presented at a nominal level (prior to amplification) of 65 dB SPL: (1) A single female talker, digitally filtered so that the long-term average spectrum matched that of the target speech; (2) a noise with the same average spectrum as the target speech, but with the temporal envelope of the single talker; (3) a noise with the same overall spectral shape as the target speech, but filtered so as to have 4 equivalent-rectangular-bandwidth (ERB) wide spectral notches at several frequencies; (4) a noise with both spectral and temporal dips obtained by applying the temporal envelope of a single talker to speech-shaped noise [as in (2)] and then filtering that noise [as in (3)]. Mean SRTs were 5-6 dB lower (better) in all of the conditions with amplification than for unaided listening. SRTs were significantly lower for the systems with one-, four-, and eight-channel compression than for linear amplification, although the benefit, averaged across subjects, was typically only 0.5 to 0.9 dB. The lowest mean SRT (-9.9 dB, expressed as a speech-to-background ratio) was obtained for noise (4) and the system with eight-channel compression. This is about 6 dB worse than for elderly subjects with near-normal hearing, when tested without amplification. It is concluded that amplification, and especially fast-acting compression amplification, can improve the ability to understand speech in background sounds with spectral and temporal dips, but it does not restore performance to normal.

Syllabic compression: Effective compression ratios for signals modulated at different rates

Compression circuits are being used increasingly in hearing aids to reduce the dynamic range of signals. Their performance is usually characterized by: (1) the threshold sound level above which the compression starts to operate; (2) the compression ratio, which is the change in input level (in dB) required to achieve a 1 dB change in output level; and (3) the attack and release times over which the signal is integrated to determine the necessary gain change. In many practical situations, the effective compression ratio obtained with dynamically varying signals such as speech is less than the compression ratio obtained using standard test signals (slow square-wave modulation with large modulation depth). This article describes the effective compression ratios achieved with sinusoidal modulation, as a function of modulation rate, level relative to the compression threshold, compression ratio and time constants. The effects of compression on a typical speech signal are also discussed.

Comparison of different forms of compression using wearable digital hearing aids

Four different compression algorithms were implemented in wearable digital hearing aids: (1) The slow-acting dual-front-end automatic gain control (AGC) system [B. C. J. Moore, B. R. Glasberg, and M. A. Stone, Br. J. Audiol. 25, 171-182 (1991)], combined with appropriate frequency response equalization, with a compression threshold of 63 dB sound pressure level (SPL) and with a compression ratio of 30 (DUAL-HI); (2) The dual-front-end AGC system combined with appropriate frequency response equalization, with a compression threshold of 55 dB SPL and with a compression ratio of 3 (DUAL-LO). This was intended to give some impression of the levels of sounds in the environment; (3) Fast-acting full dynamic range compression in four channels (FULL-4). The compression was designed to minimize envelope distortion due to overshoots and undershoots; (4) A combination of (2) and (3) above, where each applied less compression than when used alone (DUAL-4). Initial fitting was partly based on the concept of giving a flat specific-loudness pattern for a 65-dB SPL speech-shaped noise input, and this was followed by fine tuning using an adaptive procedure with speech stimuli. Eight subjects with moderate to severe cochlear hearing loss were tested in a counter-balanced design. Subjects had at least 2 weeks experience with each system in everyday life before evaluation using the Abbreviated Profile of Hearing Aid Benefit (APHAB) test and measures of speech intelligibility in quiet (AB word lists at 50 and 80 dB SPL) and noise (adoptive sentence lists in speech-shaped noise, or that same noise amplitude modulated with the envelope of speech from a single talker). The APHAB scores did not indicate clear differences between the four systems. Scores for the AB words in quiet were high for all four systems at both 50 and 80 dB SPL. The speech-to-noise ratios required for 50% intelligibility were low (indicating good performance) and similar for all the systems, but there was a slight trend for better performance in modulated noise with the DUAL-4 system than with the other systems. A subsequent trial where three subjects directly compared each of the four systems in their everyday lives indicated a slight preference for the DUAL-LO system. Overall, the results suggest that it is not necessary to compress fast modulations of the input signal.

Spectro-Temporal Characteristics of Speech at High Frequencies, and the Potential for Restoration of Audibility to People with Mild-to-Moderate Hearing Loss

Objectives: It is possible for auditory prostheses to provide amplification for frequencies above 6 kHz. However, most current hearing-aid fitting procedures do not give recommended gains for such high frequencies. This study was intended to provide information that could be useful in quantifying appropriate high-frequency gains, and in establishing the population of hearing-impaired people who might benefit from such amplification. Design: The study had two parts. In the first part, wide-bandwidth recordings of normal conversational speech were obtained from a sample of male and female talkers. The recordings were used to determine the mean spectral shape over a wide frequency range, and to determine the distribution of levels (the speech dynamic range) as a function of center frequency. In the second part, audiometric thresholds were measured for frequencies of 0.125, 0.25, 0.5, 1, 2, 3, 4, 6, 8, 10, and 12.5 kHz for both ears of 31 people selected to have mild or moderate cochlear hearing loss. The hearing loss was never greater than 70 dB for any frequency up to 4 kHz. Results: The mean spectrum level of the speech fell progressively with increasing center frequency above about 0.5 kHz. For speech with an overall level of 65 dB SPL, the mean 1/3-octave level was 49 and 37 dB SPL for center frequencies of 1 and 10 kHz, respectively. The dynamic range of the speech was similar for center frequencies of 1 and 10 kHz. The part of the dynamic range below the root-mean-square level was larger than reported in previous studies. The mean audiometric thresholds at high frequencies (10 and 12.5 kHz) were relatively high (69 and 77 dB HL, respectively), even though the mean thresholds for frequencies below 4 kHz were 41 dB HL or better. Conclusions: To partially restore audibility for a hearing loss of 65 dB at 10 kHz would require an effective insertion gain of about 36 dB at 10 kHz. With this gain, audibility could be (partly) restored for 25 of the 62 ears assessed.

Tolerable hearing aid delays. II. Estimation of limits imposed during speech production.

Objective We used real-time processing in a wearable digital hearing aid to examine the effect of processing delay on normal-hearing participants while speaking. Objective and subjective data were recorded so as to permit analysis of both the production and perception of speech read aloud from a script. We also asked participants to rate the disturbance of the echo introduced by the delay. Design Thirty-two (16M, 16F) participants were fitted binaurally with behind-the-ear (BTE) aids connected to a digital processor. A 4 mm Libby horn, surrounded by an expanding foam earplug, conducted processed sound into each ear canal. The processor provided either linear processing or three-channel, fast-acting wide dynamic range compression, independently to each ear. Insertion gains were set, using a KEMAR manikin, to be 0 dB over a wide frequency range, for frontally presented speech with a free field level of 65 dB SPL. Additionally, the aids introduced one of four selectable delays (7 to 43 msec) between the BTE microphone and receiver. After a short period of acclimatization, each participant read 16 prose passages of about 500 words in length in each of two similar-sized rooms with markedly different acoustics: reverberant and nonreverberant. For each passage, a subjective rating of the level of disturbance of the perceived echo was recorded, as well as simultaneous recordings from a microphone and a Laryngograph, which directly records glottal pulses. Results Disturbance ratings generally increased monotonically with increasing delay. Averaged results show that a delay between 25 and 30 msec is rated as “disturbing.” Measures were also taken of word production rate, speech level and range of level as well as fundamental frequency and range of fundamental frequency. For these measures of speech production, there was no significant effect until the delay exceeded 30 msec. There was little effect of acoustic environment or aid processing (linear or compression). Conclusions The acceptability of delays introduced by digital hearing aids is primarily determined by aspects of the perception of self-generated speech. Speech production, on average, is hardly affected unless the processing delay exceeds 30 msec. The permissible limit of 20 to 30 msec is smaller than the delays at which audio-visual integration is disrupted.

Development of a new method for deriving initial fittings for hearing aids with multi-channel compression: CAMEQ2-HF

Abstract 33 described a procedure, CAMEQ, for the initial fitting of multi-channel compression hearing aids. The procedure was derived using a model of loudness perception for impaired hearing. We describe here the development of a new fitting method, CAMEQ2-HF, which differs from CAMEQ in the following ways: (1) CAMEQ2-HF gives recommended gains for centre frequencies up to 10 kHz, whereas the upper limit for CAMEQ is 6 kHz; (2) CAMEQ is based on the assumption that the hearing aid user faces the person they wish to hear and uses a free-field-to-eardrum transfer function for frontal incidence. CAMEQ2-HF is based on the assumption that the user may wish to hear sounds from many directions, and uses a diffuse-field-to-eardrum transfer function; (3) CAMEQ2-HF is based on an improved loudness model for impaired hearing; (4) CAMEQ2-HF is based on recent wideband measurements of the average spectrum of speech. Sumario 33 describieron un procedimiento, CAMEQ, para la adaptación inicial de auxiliares auditivos con compresión de múltiples canales. El procedimiento se basó en el uso de un modelo de percepción de la intensidad subjetiva para casos de audición disminuida. Describimos aquí el desarrollo de un nuevo método de adaptación, CAMEQ2-HF, que difiere del CAMEQ en las siguientes formas: (1) CAMEQ2-HF proporciona ganancias recomendadas para las frecuencias centrales hasta 10 kHz mientras que el límite superior para el CAMEQ es 6 kHz; (2) CAMEQ se basa en la suposición de que el usuario de un auxiliar auditivo (se ubica de frente)hace frente a la persona que quiere oír y usa la función de transferencia campo libre-membrana timpánica para una incidencia frontal. CAMEQ2-HF se basa en la suposición de que el usuario quiere oír los sonidos desde muchas direcciones y usa una función de transferencia campodifuso-a-membrana timpánica; (3) CAMEQ2-HF se basa en un modelo de intensidad subjetiva mejorada para la audición disminuida; (4) CAMEQ2-HF se basa en las recientes mediciones de banda ancha del espectro promedio del habla.

Non-linear σ models: A perturbative approach to symmetry restoration

Abstract We examine the extent to which pertubation theory, about the naive Goldstone vacuum in two dimensions, is able to reveal the true state of affairs - that there are no Goldstone modes and that the symmetry is restored for arbitrarily small coupling constant. We exploit the recent observation, by Elitzur, of the infrared finiteness of globally invariant Green functions to compute them to third non-trivial order in a variety of models. Applying the renormalization group to improve these calculations we argue that one can see cluster decomposition beginning to restore the symmetry. The perturbation theorys ignorance of the global topology means that we cannot follow the decay of the Green function out to infinity because of a “non-perturbative barrier”. We comment on the relevance of this to “perturbative confinement” in QCD. We also examine the way in which the global topology influences the phase diagrams when we add the two-dimensional analogue of Higgs fields.

Effect of spatial separation, extended bandwidth, and compression speed on intelligibility in a competing-speech task.

The benefit for speech intelligibility of extending the bandwidth of hearing aids was assessed when the target speech (sentences) and background (two talkers) were co-located or spatially separated. Also, the relative benefits of slow and fast compression were assessed. Sixteen hearing-impaired (HI) subjects with mild-to-moderate high-frequency hearing loss and eight normal-hearing (NH) subjects were tested. The target and interfering sounds were recorded using a KEMAR manikin and were located at +/-60 degrees azimuth, either co-located or spatially separated. Simulated binaural hearing-aid processing using five-channel slow or fast compression was performed offline, with gains set individually for each HI subject. Upper cutoff frequencies were 5, 7.5, or 10 kHz. Processed stimuli were presented via headphones. For both NH (unaided) and HI subjects, there was no significant effect of cutoff frequency for the co-located condition, but a small but significant benefit from increasing the cutoff frequency from 5 to 7.5 kHz for the spatially separated condition. For the HI subjects, slow compression gave slightly but significantly higher scores than fast compression for the spatially separated but not for the co-located condition. There were marked individual differences both in the benefit from extended bandwidth and in the relative benefit of slow and fast compression.