Masking release for low- and high-pass-filtered speech in the presence of noise and single-talker interference.

Speech intelligibility was measured for sentences presented in spectrally matched steady noise, single-talker interference, or speech-modulated noise. The stimuli were unfiltered or were low-pass (LP) (1200 Hz cutoff) or high-pass (HP) (1500 Hz cutoff) filtered. The cutoff frequencies were selected to produce equal performance in both LP and HP conditions in steady noise and to limit access to the temporal fine structure of resolved harmonics in the HP conditions. Masking release, or the improvement in performance between the steady noise and single-talker interference, was substantial with no filtering. Under LP and HP filtering, masking release was roughly equal but was much less than in unfiltered conditions. When the average F0 of the interferer was shifted lower than that of the target, similar increases in masking release were observed under LP and HP filtering. Similar LP and HP results were also obtained for the speech-modulated-noise masker. The findings are not consistent with the idea that pitch conveyed by the temporal fine structure of low-order harmonics plays a crucial role in masking release. Instead, any reduction in speech redundancy, or manipulation that increases the target-to-masker ratio necessary for intelligibility to beyond around 0 dB, may result in reduced masking release.

Estimates of compression at low and high frequencies using masking additivity in normal and impaired ears.

Auditory compression was estimated at 250 and 4000 Hz by using the additivity of forward masking technique, which measures the effects on signal threshold of combining two temporally nonoverlapping forward maskers. The increase in threshold in the combined-masker condition compared to the individual-masker conditions can be used to estimate compression. The signal was a 250 or 4000 Hz tone burst and the maskers (M1 and M2) were bands of noise. Signal thresholds were measured in the presence of M1 and M2 alone and combined for a range of masker levels. The results were used to derive response functions at each frequency. The procedure was conducted with normal-hearing and hearing-impaired listeners. The results suggest that the response function in normal ears is similar at 250 and 4000 Hz with a mid level compression exponent of about 0.2. However, compression extends over a smaller range of levels at 250 Hz. The results confirm previous estimates of compression using temporal masking curves (TMCs) without assuming a linear off-frequency reference as in the TMC procedure. The impaired ears generally showed less compression. Importantly, some impaired ears showed a linear response at 250 Hz, providing a further indication that low-frequency compression originates in the cochlea.

Evaluation of companding-based spectral enhancement using simulated cochlear-implant processing.

This study tested a time-domain spectral enhancement algorithm that was recently proposed by Turicchia and Sarpeshkar [IEEE Trans. Speech Audio Proc. 13, 243-253 (2005)]. The algorithm uses a filter bank, with each filter channel comprising broadly tuned amplitude compression, followed by more narrowly tuned expansion (companding). Normal-hearing listeners were tested in their ability to recognize sentences processed through a noise-excited envelope vocoder that simulates aspects of cochlear-implant processing. The sentences were presented in a steady background noise at signal-to-noise ratios of 0, 3, and 6 dB and were either passed directly through an envelope vocoder, or were first processed by the companding algorithm. Using an eight-channel envelope vocoder, companding produced small but significant improvements in speech reception. Parametric variations of the companding algorithm showed that the improvement in intelligibility was robust to changes in filter tuning, whereas decreases in the time constants resulted in a decrease in intelligibility. Companding continued to provide a benefit when the number of vocoder frequency channels was increased to sixteen. When integrated within a sixteen-channel cochlear-implant simulator, companding also led to significant improvements in sentence recognition. Thus, companding may represent a readily implementable way to provide some speech recognition benefits to current cochlear-implant users.

A low-power asynchronous interleaved sampling algorithm for cochlear implants that encodes envelope and phase information.

Cochlear implants currently fail to convey phase information, which is important for perceiving music, tonal languages, and for hearing in noisy environments. We propose a bio-inspired asynchronous interleaved sampling (AIS) algorithm that encodes both envelope and phase information, in a manner that may be suitable for delivery to cochlear implant users. Like standard continuous interleaved sampling (CIS) strategies, AIS naturally meets the interleaved-firing requirement, which is to stimulate only one electrode at a time, minimizing electrode interactions. The majority of interspike intervals are distributed over 1-4 ms, thus staying within the absolute refractory limit of neurons, and form a more natural, pseudostochastic pattern of firing due to complex channel interactions. Stronger channels are selected to fire more often but the strategy ensures that weaker channels are selected to fire in proportion to their signal strength as well. The resulting stimulation rates are considerably lower than those of most modern implants, saving power yet delivering higher potential performance. Correlations with original sounds were found to be significantly higher in AIS reconstructions than in signal reconstructions using only envelope information. Two perceptual tests on normal-hearing listeners verified that the reconstructed signals enabled better melody and speech recognition in noise than those processed using tone-excited envelope-vocoder simulations of cochlear implant processing. Thus, our strategy could potentially save power and improve hearing performance in cochlear implant users.

Level dependence of auditory filters in nonsimultaneous masking as a function of frequency.

Auditory filter bandwidths were measured using nonsimultaneous masking, as a function of signal level between 10 and 35 dB SL for signal frequencies of 1, 2, 4, and 6 kHz. The brief sinusoidal signal was presented in a temporal gap within a spectrally notched noise. Two groups of normal-hearing subjects were tested, one using a fixed masker level and adaptively varying signal level, the other using a fixed signal level and adaptively varying masker level. In both cases, auditory filters were derived by assuming a constant filter shape for a given signal level. The filter parameters derived from the two paradigms were not significantly different. At 1 kHz, the equivalent rectangular bandwidth (ERB) decreased as the signal level increased from 10 to 20 dB SL, after which it remained roughly constant. In contrast, at 6 kHz, the ERB increased consistently with signal levels from 10 to 35 dB SL. The results at 2 and 4 kHz were intermediate, showing no consistent change in ERB with signal level. Overall, the results suggest changes in the level dependence of the auditory filters at frequencies above 1 kHz that are not currently incorporated in models of human auditory filter tuning.