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in tonal languages such as Mandarin Chinese, a lexical tone carries semantic information and is preferentially processed in the left brain hemisphere of native speakers as revealed by the functional MRI or positron emission tomography studies, which likely measure the temporally aggregated neural events including those at an attentive stage of auditory processing. Here, we demonstrate that early auditory processing of a lexical tone at a preattentive stage is actually lateralized to the right hemisphere. We frequently presented to native Mandarin Chinese speakers a meaningful auditory word with a consonant-vowel structure and infrequently varied either its lexical tone or initial consonant using an odd-ball paradigm to create a contrast resulting in a change in word meaning. The lexical tone contrast evoked a stronger preattentive response, as revealed by whole-head electric recordings of the mismatch negativity, in the right hemisphere than in the left hemisphere, whereas the consonant contrast produced an opposite pattern. Given the distinct acoustic features between a lexical tone and a consonant, this opposite lateralization pattern suggests the dependence of hemisphere dominance mainly on acoustic cues before speech input is mapped into a semantic representation in the processing stream.

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance Cl performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment 1, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall. preference. In Experiment 11, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment 1, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated Cl speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed. (c) 2006 Acoustical Society of America.

Three experiments were conducted to study relative contributions of speaking rate, temporal envelope, and temporal fine structure to clear speech perception. Experiment I used uniform time scaling to match the speaking rate between clear and conversational speech. Experiment II decreased the speaking rate in conversational speech without processing artifacts by increasing silent gaps between phonetic segments. Experiment III created "auditory chimeras" by mixing the temporal envelope of clear speech with the fine structure of conversational speech, and vice versa. Speech intelligibility in normal-hearing listeners was measured over a wide range of signal-to-noise ratios to derive speech reception thresholds (SRT). The results showed that processing artifacts in uniform time scaling, particularly time compression, reduced speech intelligibility. Inserting gaps in conversational speech improved the SRT by 1.3 dB, but this improvement might be a result of increased short-term signal-to-noise ratios during level normalization. Data from auditory chimeras indicated that the temporal envelope cue contributed more to the clear speech advantage at high signal-to-noise ratios, whereas the temporal fine structure cue contributed more at low signal-to-noise ratios. Taken together, these results suggest that acoustic cues for the clear speech advantage are multiple and distributed. (c) 2006 Acoustical Society of America.

Purpose: Speech perception in participants with auditory neuropathy (AN) was systematically studied to answer the following 2 questions: Does noise present a particular problem for people with AN? Can clear speech and cochlear implants alleviate this problem? Method: The researchers evaluated the advantage in intelligibility of clear speech over conversational speech in 13 participants with AN. Of these participants, 7 had received a cochlear implant. Eight sentence-recognition experiments were conducted to examine the clear speech advantage in 2 listening conditions (quiet and noise) using 4 stimulation modes (monaural acoustic, diotic acoustic, monaural electric, and binaurally combined acoustic and electric stimulation). Results: Participants with AN performed more poorly in speech recognition in noise than did the normal-hearing, cochlear-impaired, and cochlear implant controls. A significant clear speech advantage was observed, ranging from 9 to 23 percentage points in intelligibility for all listening conditions and stimulation modes. Electric stimulation via a cochlear implant produced significantly higher intelligibility than acoustic stimulation in both quiet and in noise. Binaural hearing with either diotic acoustic stimulation or combined acoustic and electric stimulation produced significantly higher intelligibility than monaural stimulation in quiet but not in noise. Conclusions: Participants with AN most likely derive the clear speech advantage from enhanced temporal properties in clear speech and improved neural synchrony with electric stimulation. Although the present result supports cochlear implantation as one treatment choice for people with AN, it suggests that the use of innovative hearing aids may be another viable option to improve speech perception in noise.

Natural spoken language processing includes not, only speech recognition but also identification of the speaker's gender, age, emotional, and social status. Our purpose in this study is to evaluate whether temporal cues are sufficient to support both speech and speaker recognition. Ten cochlear-implant and six normal-hearing subjects were presented with vowel tokens spoken by three men, three women, two boys, and two girls. In one condition, the subject was asked to recognize the vowel. In the other condition, the subject was asked to identify the speaker. Extensive training was provided for the speaker recognition task. Normal-hearing subjects achieved nearly perfect performance in both tasks. Cochlear-implant subjects achieved good performance in vowel recognition but poor performance in speaker recognition. The level of the cochlear implant performance was functionally equivalent to normal performance with eight spectral bands for vowel recognition but only to one band for speaker recognition. These results show a disassociation between speech and speaker recognition with primarily temporal cues, highlighting the limitation of current speech processing strategies in cochlear implants. Several methods, including explicit encoding of fundamental frequency and frequency modulation, are proposed to improve speaker recognition for current cochlear implant users. (C) 2005 Acoustical Society of America.

One reason for the poor pitch performance in current cochlear-implant users may be the highly synchronized neural firing in electric hearing that lacks stochastic properties of neural firing in normal acoustic hearing. This study used three different electric stimulation patterns, jittered, probabilistic, and auditory-model-generated pulses, to mimic some aspects of the normal neural firing pattern in acoustic hearing. Pitch discrimination was measured at standard frequencies of 100, 250, 500, and 1000 Hz on three Nucleus-24 cochlear-implant users. To test the utility of the autocorrelation pitch perception model in electric hearing, one, two, and four electrodes were stimulated independently with the same patterned electric stimulation. Results showed no improvement in performance with any experimental pattern compared to the fixed-rate control. Pitch discrimination was actually worsened with the jittered pattern at low frequencies (125 and 250 Hz) than that of the control, suggesting that externally introduced stochastic properties do not improve pitch perception in electric stimulation. The multiple-electrode stimulation did not improve performance but did not degrade performance either. The present results suggest that both "the right time and the right place" may be needed to restore normal pitch perception in cochlear-implant users. (c) 2005 Acoustical Society of America.

Speech recognition in noise and music perception is especially challenging for current cochlear implant users. The present study utilizes the residual acoustic hearing in the nonimplanted ear in five cochlear implant users to elucidate the role of temporal fine structure at low frequencies in auditory perception and to test the hypothesis that combined acoustic and electric hearing produces better performance than either mode alone. The first experiment measured speech recognition in the presence of competing noise. It was found that, although the residual low-frequency (< 1000 Hz) acoustic hearing produced essentially no recognition for speech recognition in noise, it significantly enhanced performance when combined with the electric hearing. The second experiment measured melody recognition in the same group of subjects and found that, contrary to the speech recognition result, the low-frequency acoustic hearing produced significantly better performance than the electric hearing. It is hypothesized that listeners with combined acoustic and electric hearing might use the correlation between the salient pitch in low-frequency acoustic hearing and the weak pitch in the envelope to enhance segregation between signal and noise. The present study suggests the importance and urgency of accurately encoding the fine-structure cue in cochlear implants. (c) 2005 Acoustical Society of America.

Amplitude modulation (AM) and frequency modulation (FM) are commonly used in communication, but their relative contributions to speech recognition have not been fully explored. To bridge this gap, we derived slowly varying AM and FM from speech sounds and conducted listening tests using stimuli with different modulations in normal-hearing and cochlear-implant subjects. We found that although AM from a limited number of spectral bands may be sufficient for speech recognition in quiet, FM significantly enhances speech recognition in noise, as well as speaker and tone recognition. Additional speech reception threshold measures revealed that FM is particularly critical for speech recognition with a competing voice and is independent of spectral resolution and similarity. These results suggest that AM and FM provide independent yet complementary contributions to support robust speech recognition under realistic listening situations. Encoding FM may improve auditory scene analysis, cochlear-implant, and audiocoding performance.

Different from traditional Fourier analysis, a signal can be decomposed into amplitude and frequency modulation components. The speech processing strategy in most modern cochlear implants only extracts and encodes amplitude modulation in a limited number of frequency bands. While amplitude modulation encoding has allowed cochlear implant users to achieve good speech recognition in quiet, their performance in noise is severely compromised. Here, we propose a novel speech processing strategy that encodes both amplitude and frequency modulations in order to improve cochlear implant performance in noise. By removing the center frequency from the subband signals and additionally limiting the frequency modulation's range and rate, the present strategy transforms the fast-varying temporal fine structure into a slowly varying frequency modulation signal. As a first step, we evaluated the potential contribution of additional frequency modulation to speech recognition in noise via acoustic simulations of the cochlear implant. We found that while amplitude modulation from a limited number of spectral bands is sufficient to support speech recognition in quiet, frequency modulation is needed to support speech recognition in noise. In particular, improvement by as much as 71 percentage points was observed for sentence recognition in the presence of a competing voice. The present result strongly suggests that frequency modulation be extracted and encoded to improve cochlear implant performance in realistic listening situations. We have proposed several implementation methods to stimulate further investigation.

Frequency modulation (FM) detection was investigated in acoustic and electric hearing to characterize cochlear-implant subjects' ability to detect dynamic frequency changes and to assess the relative contributions of temporal and spectral cues to frequency processing. Difference limens were measured for frequency upward sweeps, downward sweeps, and sinusoidal FM as a function of standard frequency and modulation rate. In electric hearing, factors including electrode position and stimulation level were also studied. Electric hearing data showed that the difference limen increased monotonically as a function of standard frequency regardless of the modulation type, the modulation rate. the electrode position, and the stimulation level. In contrast, acoustic hearing data showed that the difference limen was nearly a constant as a function of standard frequency. This difference was interpreted to mean that temporal cues are used only at low standard frequencies and at low modulation rates. At higher standard frequencies and modulation rates, the reliance on the place cue is increased, accounting for the better performance in acoustic hearing than for electric hearing with single-electrode stimulation. The present data suggest a speech processing strategy that encodes slow frequency changes using lower stimulation rates than those typically employed by contemporary cochlear-implant speech processors. (C) 2004 Acoustical Society of America.

When instructed to speak clearly for people with hearing loss, a talker can effectively enhance the intelligibility of his/her speech by producing "clear" speech. We analyzed global acoustic properties of clear and conversational speech from two talkers and measured their speech intelligibility over a wide range of signal-to-noise ratios in acoustic and electric hearing. Consistent with previous studies, we, found that clear speech had a slower overall rate, higher temporal amplitude modulations, and also produced higher intelligibility than conversational speech. To delineate the role of temporal amplitude modulations in clear speech, we extracted the temporal envelope from a number of frequency bands and replaced speech fine-structure with noise fine-structure to simulate cochlear implants. Although both simulated and actual cochlear-implant listeners required higher signal-to-noise ratios to achieve normal performance, a 3-4 dB difference in speech reception threshold was preserved between clear and conversational speech for all experimental conditions. These results suggest that while temporal fine structure is important for speech recognition in noise in general, the temporal envelope carries acoustic cues that contribute to the clear speech intelligibility advantage. (C) 2004 Acoustical Society of America.

It is important to know what cues the sensory system extracts from natural stimuli and how the brain uses them to form perception. To explore this issue, Smith, Delgutte, and Oxenham [Nature (London) 416, 87-90 (2002)] mixed one sound's temporal envelope with another sound's fine temporal structure to produce auditory chimaeras and found that "the perceptual importance of the envelope increases with the number of frequency bands, while that of the fine structure diminishes." This study addressed two technical issues related to natural cochlear filtering and artificial filter ringing in the chimaerizing algorithm. In addition, this study found that the dichotomy in auditory perception revealed by auditory chimaeras is an epiphenomenon of the classic dichotomy between low- and high-frequency processing. Finally, this study found that the temporal envelope determines sound location as long as the interaural level difference cue is present. The present result reinforces the original hypothesis that the temporal envelope is critical for speech perception whereas temporal fine structure is critical for pitch perception, but does not support the assertion regarding the temporal envelope and fine structure as the acoustic basis for the "what" and "where" mechanisms. (C) 2004 Acoustical Society of America.

Hearing aids and cochlear implants are two major hearing enhancement technologies but yet share little in research and development. The purpose of this study was to determine whether hearing aid directional microphones and noise reduction technologies could enhance cochlear implant users' speech understanding and ease of listening. Digital hearing aids serving as preprocessors were programmed to omni-directional microphone, directional microphone, and directional microphone plus noise reduction conditions. Three groups of subjects were tested with the hearing aid processed speech stimuli. Results indicated that hearing aids with directional microphones and noise reduction algorithms significantly enhanced speech understanding and listening comfort. (C) 2004 Acoustical Society of America.

The present study aimed to examine the effect of noise on vowel-like suprathreshold discrimination in cochlear-implant listeners. The task was to detect an increment in level at the middle harmonic (400 Hz) in the background of a seven-harmonic complex from 100 to 700 Hz in 100-Hz steps. The task was performed in the absence (control) and presence of a white noise presented over a 20-35-dB range from inaudible to loud. The present result shows that discrimination of suprathreshold harmonic stimuli was significantly enhanced, particularly at the soft signal level, with suprathreshold noise. This result suggests that tuning of the noise level is required to optimize performance of different tasks in cochlear implants. (C) 2003 Elsevier B.V. All rights reserved.

Xishan coalfield, Shanxi, is located in the northwest of the Qinshui Basin, central North China. It is notable for its varieties of coal rank ranging from high volatile bituminous coals to anthracite as well as having abundant coalbed methane resources. Zircon fission track (ZFT) analyses were carried out on the zircons in 2 Upper Carboniferous and 5 Lower‐Middle Permian sandstones, and vitrinite reflectance of Late Carboniferous and Early Permian coals were measured to determine the timing of thermal events and maximum paleo‐temperatures, which were responsible for coal maturation and coalbed methane generation. Maximum paleo‐temperatures calculated from vitrinite reflectance values reached to about 232 and 223 °C in Late Carboniferous and Early Permian coals, respectively, and the estimated paleo‐temperature gradient was 11.84 °C/100 m, representing an intensive abnormal thermal event. Results of the ZFT dating indicated that 5 samples failed the χ2‐test and 2 samples passed the test. The decomposition results of the 5 samples divided their age populations into 3 periods: (a) older ages (537, 584, and 802 Ma) than sandstones ages, (b) close to or slightly older than their depositional ages (289, 301, and 331 Ma), and (c) younger than the depositional ages (181–215). The 2 samples that passed χ2‐test yield the central ages of 168 ± 7 Ma and 190 ± 8 Ma, respectively, younger than the deposition age. The close to or older ages than the sandstones depositional ages represent the tectonothermal events occurring in their source areas; the younger ages indicate the existence of the postdepositional tectonothermal event. The agreement of the partly annealing temperature zone (210–300 °C) of zircon fission tracks with the calculated maximum paleo‐temperatures from vitrinite reflectance suggests a Late Triassic‐Early Jurassic abnormal thermal event with the formation time of the present coal rank being 181–215 Ma, rather than a unique intrusion at 95–135 Ma on the western margin of coalfield as previously believed. Combined with other ZFT ages regionally, this abnormal event also occurred in the southern as well as the northern parts of the Qinshui Basin. The Late Triassic‐Early Jurassic intensive extension in the North China Craton is the geodynamic setting of this tectonothermal event. [ABSTRACT FROM AUTHOR]

The Palaeo-Asian Ocean (PAO) was located between Siberia and the North China Craton (NCC). The time of its closure, as well as the crustal architecture of the surrounding region, are still uncertain. In this paper, we address these problems using LA-ICP-MS U-Pb ages of zircon grains in three magmatic rocks and three river sand samples of the Guyang area at the northern margin of the NCC. The magmatic rocks yield early Carboniferous crystallization ages of 340.3 ± 7.5 Ma, 331.7 ± 2.3 Ma and 341.1 ± 1.9 Ma for the main granitic pluton, a granodiorite dyke and a diorite enclave, respectively. They were the product of subduction of the PAO underneath the NCC. The detrital zircon U-Pb ages in the river sand samples are dominated by two groups at 2550-2400 and 290-260 Ma, with subordinate groups at 2700-2600, 2000-1800 and 320-300 Ma, dispersed Early Palaeoproterozoic grains and a minor but important group of Cretaceous grains in sands of the Aibugai River. These results, combined with previous geochronological data, permit a re-assessment of the tectonic evolution of the northern margin of the NCC. The events at ~2.5 Ga reflect major crustal reworking and cratonization of the northern margin of the NCC, which then became a stable platform lasting until the Carboniferous. The period between 2.0 and 1.8 Ga was characterized by major collisions related to formation of the Khondalite Belt and the Trans-North China Orogen. In the Carboniferous, subduction of the PAO underneath the northern margin of the NCC developed a zoned distribution of magmatic rocks. The PAO is now constrained to have closed in the Early Permian by the collision of the NCC and Siberia. The youngest group of Cretaceous zircons reflects magmatic events related to lithosphere thinning beneath the Eastern Block of the NCC. Copyright © 2016 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]

Good performance in cochlear implant users depends in large part on the ability of a speech processor to effectively decompose speech signals into multiple channels of narrow-band electrical pulses for stimulation of the auditory nerve. Speech processors that extract only envelopes of the narrow-band signals (e.g., the continuous interleaved sampling (CIS) processor) may not provide sufficient information to encode the tonal cues in languages such as Chinese. To improve the performance in cochlear implant users who speak tonal language, we proposed and developed a novel speech-processing strategy, which extracted both the envelopes of the narrow-band signals and the fundamental frequency (F0) of the speech signal, and used them to modulate both the amplitude and the frequency of the electrical pulses delivered to stimulation electrodes. We developed an algorithm to extract the fundatmental frequency and identified the general patterns of pitch variations of four typical tones in Chinese speech. The effectiveness of the extraction algorithm was verified with an artificial neural network that recognized the tonal patterns from the extracted F0 information. We then compared the novel strategy with the envelope-extraction CIS strategy in human subjects with normal hearing. The novel strategy produced significant improvement in perception of Chinese tones, phrases, and sentences. This novel processor with dynamic modulation of both frequency and amplitude is encouraging for the design of a cochlear implant device for sensorineurally deaf patients who speak tonal languages.

Auditory neuropathy affects the normal synchronous activity in the auditory nerve, without affecting the amplification function in the inner ear. Patients with auditory neuropathy often complain that they can hear sounds, but cannot understand speech. Here we report psychophysical tests indicating that these patients' poor speech recognition is due to a severe impairment in their temporal processing abilities. We also simulate this temporal processing impairment in normally hearing listeners and produce similar speech recognition deficits. This study demonstrates the importance of neural synchrony for auditory perceptions including speech recognition in humans. The results should contribute to better diagnosis and treatment of auditory neuropathy.

Psychophysical laws relate the intensity of a physical stimulus to its perceived magnitude. G.T. Fechner hypothesized 150 years ago that the psychophysical law can be derived by measuring intensity discrimination, but modern scientists favor a direct magnitude estimation approach and are still divided on whether and how intensity discrimination is related to sensation magnitude. This controversy is partially due to the uncertainty of the role of the sensory organ in the psychophysical law. Here we bypass the auditory sensory organ with electric stimulation of the human auditory nerve and find a close coupling between intensity discrimination and loudness functions in electric hearing. Our results support Fechner's hypothesis in principle but not the exact relationship from which he derived his logarithmic law.

Recognition of consonants, vowels, and sentences was measured in conditions of reduced spectral resolution and distorted spectral distribution of temporal envelope cues. Speech materials were processed through four bandpass filters (analysis bands), half-wave rectified, and low-pass filtered to extract the temporal envelope from each band. The envelope from each speech band modulated a band-limited noise (carrier bands). Analysis and carrier bands were manipulated independently to alter the spectral distribution of envelope cues. Experiment I demonstrated that the location of the cutoff frequencies defining the bands was not a critical parameter for speech recognition, as long as the analysis and carrier bands were matched in frequency extent. Experiment II demonstrated a dramatic decrease in performance when the analysis and carrier bands did not match in frequency extent, which resulted in a warping of the spectral distribution of envelope cues. Experiment III demonstrated a large decrease in performance when the carrier bands were shifted in frequency, mimicking the basal position of electrodes in a cochlear implant. And experiment IV showed a relatively minor effect of the overlap in the noise carrier bands, simulating the overlap in neural populations responding to adjacent electrodes in a cochlear implant. Overall, these results show that, for four bands, the frequency alignment of the analysis bands and carrier bands is critical for good performance, while the exact frequency divisions and overlap in carrier bands are not as critical.

Recent studies have shown that temporal waveform envelope cues can provide significant information for English speech recognition. This study investigated the use of temporal envelope cues in a tonal language: Mandarin Chinese. In this study, the speech was divided into several frequency analysis bands; the amplitude envelope was extracted from each band by half-wave rectification and low-pass filtering and was used to modulate a noise of the same bandwidth as the analysis band. These manipulations preserved temporal and amplitude cues in each frequency band, but removed the spectral detail within each band. Chinese vowels, consonants, tones and sentences were identified by 12 native Chinese-speaking listeners with 1, 2, 3, and 4 noise bands. The results showed that the recognition score of vowels, consonants, and sentences increased monotonically with the number of bands, a pattern similar to that observed in English speech recognition. In contrast, tones were consistently recognized at about 80% correct level, independent of the number of bands. This high level of tone recognition produced a significant difference in the open-set sentence recognition between Chinese (11.0%) and English (2.9%) for the one-band condition where no spectral information was available. The data also revealed that, with primarily temporal cues, the falling-rising tone (tone 3) and the falling tone (tone 4) were more easily recognized than the flat tone (tone 1) and the rising tone (tone 2). This differential pattern in tone recognition resulted in a similar pattern in word recognition: words having either tone 3 or 4 were more likely to be recognized while words having tone 1 and 2 were not. The quantitative role of tones in Chinese speech recognition was further explored using a power-function model and found to play a significant role in relating phoneme recognition to sentence recognition.

Electric charge has long been hypothesized to be the effective stimulus variable that determines loudness evoked by directly stimulating the auditory nerve. This 'equal-charge, equal-loudness' hypothesis predicts that stimulus amplitude and duration can be traded linearly to produce equal loudness. Loudness sensations from threshold to maximum loudness were measured systematically as a function of stimulus amplitude and duration in cochlear implant listeners. The measured data do not support the equal-charge, equal-loudness hypothesis: an increment in stimulus amplitude produces a significantly louder sensation than the same change in stimulus duration. Instead of the linear equal-charge model, a power-function model successfully predicts the measured data and should be used to encode loudness in electric hearing.

Intensity coding mechanisms are explored in a paradigm involving both forward and simultaneous masking. For intensity discrimination of 1000-Hz pure tone in quiet, a near-miss to Weber's law is observed. However, as more stimulus components are added to this relatively simple experiment, interactions among components produce a more complex pattern of results. An intense forward masker, while not causing any threshold shift for the test tone, produces a nonmonotonic intensity discrimination function ["the midlevel hump," Zeng et al., Hearing Res. 55, 223-230 (1991)]. The midlevel hump can be removed by the presence of additional notched noise [Plack and Viemeister, J. Acoust. Soc. Am. 92, 1902-1910 (1992)] or narrow-band noise whose level is increased along with the test tone's standard level. The same midlevel hump can also be enhanced by a fixed-low-level notched noise or a high-level, high-pass noise which causes minimal masking at the test frequency. Interactions of forward masking and simultaneous masking present a serious problem for a clear interpretation of these results. For example, the notched noise was originally intended to restrict off-frequency listening, but on-frequency masking compromised this original purpose and confounded the interpretation of the notched noise effects. By measuring systematically the growth-of-masking functions, the present study identified various interactions of forward and simultaneous masking and clarified the role of off-frequency listening in forward-masked intensity discrimination. Both peripheral and central mechanisms may have contributed to the occurrence, reduction and enhancement of the midlevel hump under these masking conditions.

Widespread application of cochlear implants is limited by cost, especially in developing countries. In this article we present a design for a low-cost but effective cochlear implant system. The system includes a speech processor, four pairs of transmitting and receiving coils, and an electrode array with four monopolar electrodes. All implanted components are passive, reducing to a minimum the complexity of manufacture and allowing high reliability. A four-channel continuous interleaved sampling strategy is used for the speech processor. The processor and transmission link have been evaluated in tests with a subject previously implanted with the Ineraid electrode array and percutaneous connector. A prototype of the link, consisting of four pairs of transmitting and external receiving coils, was used, with the outputs of the receiving coils directed to four intracochlear electrodes through the percutaneous connector. The subject achieved speech reception scores with the prototype system that were equivalent to those achieved with a standard laboratory implementation of a continuous interleaved sampling processor with current-controlled stimuli.

Traditional loudness models have been based on the average energy and the critical band analysis of steady-state sounds. However, most environmental sounds, including speech, are dynamic stimuli, in which the average level [e.g., the root-mean-square (rms) level] does not account for the large temporal fluctuations. The question addressed here was whether two stimuli of the same rms level but different peak levels would produce an equal loudness sensation. A modern adaptive procedure was used to replicate two classic experiments demonstrating that the sensation of "beats" in a two- or three-tone complex resulted in a louder sensation [E. Zwicker and H. Fastl, Psychoacoustics-Facts and Models (Springer-Verlag, Berlin, 1990)]. Two additional experiments were conducted to study exclusively the effects of the temporal envelope on the loudness sensation of dynamic stimuli. Loudness balance was performed by normal-hearing listeners between a white noise and a sinusoidally amplitude-modulated noise in one experiment, and by cochlear implant listeners between two harmonic stimuli of the same magnitude spectra, but different phase spectra, in the other experiment. The results from both experiments showed that, for two stimuli of the same rms level, the stimulus with greater temporal fluctuations sometimes produced a significantly louder sensation, depending on the temporal frequency and overall stimulus level. In normal-hearing listeners, the louder sensation was produced for the amplitude-modulated stimuli with modulation frequencies lower than 400 Hz, and gradually disappeared above 400 Hz, resulting in a low-pass filtering characteristic which bore some similarity to the temporal modulation transfer function. The extent to which loudness was greater was a nonmonotonic function of level in acoustic hearing and a monotonically increasingly function in electric hearing. These results suggest that the loudness sensation of a dynamic stimulus is not limited to a 100-ms temporal integration process, and may be determined jointly by a compression process in the cochlea and an expansion process in the brain. A level-dependent compression scheme that may better restore normal loudness of dynamic stimuli in hearing aids and cochlear implants is proposed.

Nearly perfect speech recognition was observed under conditions of greatly reduced spectral information. Temporal envelopes of speech were extracted from broad frequency bands and were used to modulate noises of the same bandwidths. This manipulation preserved temporal envelope cues in each band but restricted the listener to severely degraded information on the distribution of spectral energy. The identification of consonants, vowels, and words in simple sentences improved markedly as the number of bands increased; high speech recognition performance was obtained with only three bands of modulated noise. Thus, the presentation of a dynamic temporal pattern in only a few broad spectral regions is sufficient for the recognition of speech.

In a earlier study we showed that loudness function depends on stimulus frequency for simple sinusoidal and pulsatile stimuli in cochlear implants. Loudness is an exponential function of stimulus amplitude for high frequencies (> 300 Hz) and a power function for low frequencies (< 300 Hz). Two experiments were conducted to extend our previous work in eight Ineraid cochlear implant subjects. First, loudness functions were measured by means of a magnitude estimation technique for simple sinusoidal stimuli. Stimuli were 100- and 1,000-Hz sinusoids. Stimuli had a duration of 300 milliseconds and were presented at amplitudes representing 10%, 30%, 50%, 70%, and 90% of the dynamic range. Loudness estimates were best fit by an exponential function for the 1,000-Hz sinusoid and by a power function for the 100-Hz sinusoid. The estimation result is consistent with previous results that were obtained with a loudness balance technique. In a second experiment, loudness functions were studied for modulated stimuli, in which a 1,000-Hz sinusoid was modulated by a 100-Hz sinusoid at either 50% or 100% modulation depth. A linear loudness-balance function was obtained between the modulated stimuli and a 1,000-Hz sinusoidal standard, indicating that the modulated stimuli have the same exponential loudness function as the 1,000-Hz carrier despite the 100-Hz modulator. This result has important implications for speech processor design, because many devices use low-frequency speech envelopes to modulate a high-frequency carrier. This result also provides a psychophysical basis for the logarithmic amplitude transformation in commonly used speech processors to compress the wide acoustic dynamic range into a narrow electric dynamic range while preserving normal loudness function.

China has approximately 6 million totally deaf people according to an official survey conducted in 1990, although the actual number is probably higher. A primary cause of deafness is the use of ototoxic drugs. There does not appear to be any emergent deaf culture in China at present. As the only available medical device that can restore partial hearing to a totally deaf person, the cochlear implant has been in development in China since 1979. This paper provides an overview of cochlear implants in China and is based on a review of published materials, visits to research institutes and hospitals, and personal communication with Chinese colleagues. As of 1993, about 1,000 deaf people, including 50 children below age 12 years, have received four types of single-electrode cochlear implants that were developed and fabricated by institutions in China. These single-electrode devices have provided an aid to lip reading, but are no longer in use due to their inability to produce open-set speech recognition. Present implant research in China focuses on development of multi-electrode devices. Basic research in electrical stimulation is relatively lacking and standardized audiological evaluation for cochlear implant effectiveness needs to be developed. The present economic growth and legal system reform in China, combined with advances in implant technology, may make it possible to produce an affordable yet effective cochlear implant system. This paper discusses cochlear implants only in China, but the social and economic factors are similar in many developing countries in Asia, South America, Eastern Europe, and Africa, where a low-cost, high-performance cochlear implant system is also needed.

The growth of loudness of a tone burst following an intense forward masker was measured as a function of the tone level. The level of the forward-masked tone was adjusted to balance the loudness of a standard tone presented without a forward masker, using a 2AFC, double-staircase, tracking procedure. The forward masker was a 90-dB SPL, 100-ms, 1000-Hz pure tone. The standard tone and the masked tone were both 25-ms, 1000-Hz pure tones. The forward masker and the masked tone were always presented in the first interval. With a 100-ms delay between them, there was little or no threshold elevation for the masked tone. However, the masker caused the masked tone to sound louder than it would if it had not been masked, a phenomenon termed "loudness enhancement" [Irwin and Zwislocki, Percept. Psychophys. 10, 189-192 (1971); Galambos et al., J. Acoust. Soc. Am. 52, 1127-1130 (1972)]. In addition, the present results show a nonmonotonic enhancement function that the forward masker introduced a 10-16-dB enhancement effect for tones of 40-65 dB SPL and no significant effect for the 30 and 90 dB SPL tones. The loudness variability in forward masking was estimated from the upper and lower sequences tracking the 21% and the 79% louder response levels on the psychometric function, respectively. The variability demonstrated a similar nonmonotonic function. In forward masking loudness grows more steeply at low-medium sensation levels, and merges with normal growth at high levels.(ABSTRACT TRUNCATED AT 250 WORDS)

Two distinct physiological mechanisms underlying loudness sensation were inferred from electric stimulation of the human auditory nerve and brainstem. In contrast to a power function relating loudness and stimulus intensity in acoustic hearing, loudness in electric stimulation of the auditory nerve depends on stimulus frequency. Loudness is an exponential function of electric amplitude for high frequencies and is a power function for low frequencies. A frequency-dependent, two-stage model is suggested to explain the loudness function, in which the first stage of processing is performed by a mechanical mechanism in the cochlea for high-frequency stimuli and by a neural mechanism in the cochlear nucleus for low-frequency stimuli.

Frequency difference limens for pure tones preceded by a forward masker or followed by a backward masker were obtained across a wide range of signal levels. Relkin and Doucet [Hear. Res. 55, 215-222 (1991)] have shown that at a masker-signal delay of 100 ms, the thresholds of high-SR (spontaneous rate) auditory-nerve fibers are recovered, while the low-SR fiber thresholds are not. Therefore, forward-masked frequency discrimination potentially offers a method to investigate the role of low-SR fibers in the coding of frequency. It has been shown that when an intense forward masker is presented 100 ms before a pure-tone signal, intensity difference limens are elevated for mid-level signals [Zeng et al., Hear. Res. 55, 223-230 (1991)]. However, Plack and Viemeister [J. Acoust. Soc. Am. 92, 3097-3101 (1992)] have shown that a similar elevation in the intensity difference limen is obtained under conditions of backward masking, where selective adaptation of the auditory neurons would not be expected to occur. A condition of backward-masked frequency discrimination was therefore included to investigate the role of interference resulting from adding additional stimuli to a discrimination task. For signals at 1000 and 6000 Hz, there was no effect of a forward masker upon frequency difference limens. For the backward-masked conditions, an elevation of the frequency difference limen was observed at all signal levels, demonstrating that the effects of forward and backward maskers upon frequency discrimination are dissimilar and suggesting that cognitive effects are present in backward-masked discrimination tasks.(ABSTRACT TRUNCATED AT 250 WORDS)

A nonmonotonic intensity discrimination function was recently reported in which a midlevel hump occurred for 25-ms sinusoidal standards ranging from 20 to 100 dB SPL and presented 100 ms after an intense narrow-band noise forward masker [F.-G. Zeng et al., Hear. Res. 55, 223-230 (1991)]. This paper provides additional data on how the midlevel hump is affected by three factors of forward masking: signal delay, masker level, and frequency. Specifically, just-noticeable differences (jnd's) in intensity were obtained at signal delays of 50, 200, and 400 ms. Results show that at the midlevels the forward-masked intensity jnd's did not recover to their unmasked values, even at the 400-ms signal delay. The longer the signal delay, the smaller this midlevel hump. This slow recovery of the midlevel jnd's is consistent with the finding that low-spontaneous rate (SR) neurons have a slow recovery from forward masking [E. M. Relkin and J. R. Doucet, Hear. Res. 55, 215-222 (1991)]. The large midlevel effect decreased sharply as masker level was reduced from 90 to 60 dB SPL, and disappeared for masker levels less than 40 dB SPL. A frequency selectivity effect for the large midlevel jnd effect was also observed, as maskers with frequency components 2 to 3 oct away from the signal frequency did not affect the jnd's. Overall, the present data are consistent with the hypothesis of Zeng et al. (1991) that low-SR neurons are involved in the midlevel hump of intensity discrimination in forward masking.

Binaural loudness balance between electric and acoustic stimulation is obtained in auditory brainstem implant listeners who had substantial acoustic hearing in one ear. The data are well described by a linear relationship between acoustic decibels and electric microamps. Based upon this linear relationship, we propose an exponential model of loudness growth in electric stimulation. The exponential model predicts that the loudness growth function can be determined solely by the threshold and the uncomfortable loudness level in electric stimulation. This prediction is consistent with previous psychophysical data on loudness functions. Implications of this finding for speech processor designs are discussed.

We obtained just-noticeable differences (jnds) for the intensity of pure tones following a forward masker. The masker was a 100-ms burst of narrow-band noise centered at 1000 Hz presented at 90 dB SPL; the pure-tone signal was at 1000 Hz and was 25 ms in duration. The masker-signal delay was 100 ms. Under these conditions, there is no threshold shift for the detection of the pure-tone signal following the forward masker. In contrast with the absence of a forward-masker effect upon detection thresholds, unusually large midlevel (40-60 dB SPL) jnds were observed. These large midlevel jnds were measured as a function of signal delay, revealing that they are not completely recovered to the normal (unmasked) values by 400 ms. We interpret these data as a consequence of the slower recovery of low-spontaneous rate, high-threshold neurons following prior stimulation (Relkin and Doucet, 1990). These experiments may therefore provide psychophysical evidence that the low-spontaneous rate, high-threshold neurons are a necessary physiological component in the coding of the large dynamic range for intensity. In addition, the present data provide evidence that the assumption that the effect of forward masking is limited to 100-200 ms is inappropriate, as this recovery time does not necessarily apply to suprathreshold tasks.

Binaural loudness matching data using a 21FC adaptive procedure were obtained in high-frequency, unilateral cochlear-impaired listeners. The matches were obtained at frequencies where both ears had similarly normal thresholds, and also at other frequencies where the impaired ear had various degrees of hearing loss. In these listeners, one presumed difference between the ears is the limited or altered spread of excitation in the impaired ear. In agreement with previous studies using other approaches (Hellman, 1974, 1978; Hellman & Meiselman, 1986; Moore, Glasberg, Hess & Birchall, 1985; Schneider & Parker, 1987), the results of the present study suggest that both the range and the slope of loudness growth function are not dependent on the spread of excitation, but instead are related primarily to the degree of threshold elevation at the test frequency. Following this suggestion, a spread-of-excitation-independent model, based upon a group of neurons with the same characteristic frequency (CF) but different thresholds, is proposed to account for loudness growth in both normal and recruitment cases. In particular, it is shown quantitatively that a compressed distribution of thresholds due to threshold elevation may be responsible for loudness recruitment in sensorineural hearing loss.

The purpose of this study was to investigate the sufficient perceptual cues used in the recognition of four voiceless fricative consonants [s, f, theta, integral of] followed by the same vowel [i:] in normal-hearing and hearing-impaired adult listeners. Subjects identified the four CV speech tokens in a closed-set response task across a range of presentation levels. Fricative syllables were either produced by a human speaker in the natural stimulus set, or generated by a computer program in the synthetic stimulus set. By comparing conditions in which the subjects were presented with equivalent degrees of audibility for individual fricatives, it was possible to isolate the factor of lack of audibility from that of loss of suprathreshold discriminability. Results indicate that (a) the friction burst portion may serve as a sufficient cue for correct recognition of voiceless fricatives by normal-hearing subjects, whereas the more intense CV transition portion, though it may not be necessary, can also assist these subjects to distinguish place information, particularly at low presentation levels; (b) hearing-impaired subjects achieved close-to-normal recognition performance when given equivalent degrees of audibility of the frication cue, but they obtained poorer-than-normal performance if only given equivalent degrees of audibility of the transition cue; (c) the difficulty that hearing-impaired subjects have in perceiving fricatives under normal circumstances may be due to two factors: the lack of audibility of the frication cue and the loss of discriminability of the transition cue.