Charissa R. Lansing

Blind estimation of reverberation time

The reverberation time (RT) is an important parameter for characterizing the quality of an auditory space. Sounds in reverberant environments are subject to coloration. This affects speech intelligibility and sound localization. Many state-of-the-art audio signal processing algorithms, for example in hearing-aids and telephony, are expected to have the ability to characterize the listening environment, and turn on an appropriate processing strategy accordingly. Thus, a method for characterization of room RT based on passively received microphone signals represents an important enabling technology. Current RT estimators, such as Schroeders method, depend on a controlled sound source, and thus cannot produce an online, blind RT estimate. Here, a method for estimating RT without prior knowledge of sound sources or room geometry is presented. The diffusive tail of reverberation was modeled as an exponentially damped Gaussian white noise process. The time-constant of the decay, which provided a measure of the RT, was estimated using a maximum-likelihood procedure. The estimates were obtained continuously, and an order-statistics filter was used to extract the most likely RT from the accumulated estimates. The procedure was illustrated for connected speech. Results obtained for simulated and real room data are in good agreement with the real RT values.

Word identification and eye fixation locations in visual and visual-plus-auditory presentations of spoken sentences

In this study, we investigated where people look on talkers’ faces as they try to understand what is being said. Sixteen young adults with normal hearing and demonstrated average speechreading proficiency were evaluated under two modality presentation conditions: vision only versus vision plus lowintensity sound. They were scored for the number of words correctly identified from 80 unconnected sentences spoken by two talkers. The results showed two competing tendencies: an eye primacy effect that draws the gaze to the talker’s eyes during silence and an information source attraction effect that draws the gaze to the talker’s mouth during speech periods. Dynamic shifts occur between eyes and mouth prior to speech onset and following the offset of speech, and saccades tend to be suppressed during speech periods. The degree to which the gaze is drawn to the mouth during speech and the degree to which saccadic activity is suppressed depend on the difficulty of the speech identification task. Under the most difficult modality presentation condition, vision only, accuracy was related to average sentence difficulty and individual proficiency in visual speech perception, but not to the proportion of gaze time directed toward the talker’s mouth or toward other parts of the talker’s face.

Binaural signal processing using multiple acoustic sensors and digital filtering

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

A two-microphone dual delay-line approach for extraction of a speech sound in the presence of multiple interferers

This paper describes algorithms for signal extraction for use as a front-end of telecommunication devices, speech recognition systems, as well as hearing aids that operate in noisy environments. The development was based on some independent, hypothesized theories of the computational mechanics of biological systems in which directional hearing is enabled mainly by binaural processing of interaural directional cues. Our system uses two microphones as input devices and a signal processing method based on the two input channels. The signal processing procedure comprises two major stages: (i) source localization, and (ii) cancellation of noise sources based on knowledge of the locations of all sound sources. The source localization, detailed in our previous paper [Liu et al., J. Acoust. Soc. Am. 108, 1888 (2000)], was based on a well-recognized biological architecture comprising a dual delay-line and a coincidence detection mechanism. This paper focuses on description of the noise cancellation stage. We designed a simple subtraction method which, when strategically employed over the dual delay-line structure in the broadband manner, can effectively cancel multiple interfering sound sources and consequently enhance the desired signal. We obtained an 8-10 dB enhancement for the desired speech in the situations of four talkers in the anechoic acoustic test (or 7-10 dB enhancement in the situations of six talkers in the computer simulation) when all the sounds were equally intense and temporally aligned.

Performance of time- and frequency-domain binaural beamformers based on recorded signals from real rooms

Extraction of a target sound source amidst multiple interfering sound sources is difficult when there are fewer sensors than sources, as is the case for human listeners in the classic cocktail-party situation. This study compares the signal extraction performance of five algorithms using recordings of speech sources made with three different two-microphone arrays in three rooms of varying reverberation time. Test signals, consisting of two to five speech sources, were constructed for each room and array. The signals were processed with each algorithm, and the signal extraction performance was quantified by calculating the signal-to-noise ratio of the output. A frequency-domain minimum-variance distortionless-response beamformer outperformed the time-domain based Frost beamformer and generalized sidelobe canceler for all tests with two or more interfering sound sources, and performed comparably or better than the time-domain algorithms for tests with one interfering sound source. The frequency-domain minimum-variance algorithm offered performance comparable to that of the Peissig-Kollmeier binaural frequency-domain algorithm, but with much less distortion of the target signal. Comparisons were also made to a simple beamformer. In addition, computer simulations illustrate that, when processing speech signals, the chosen implementation of the frequency-domain minimum-variance technique adapts more quickly and accurately than time-domain techniques.

Acoustic scene analysis using estimated impulse responses

Pre-processing for hearing aids, such as adaptive beamforming, is sensitive to characteristics of the acoustic environment, in particular, reverberation. We conjecture that knowledge of the acoustic environment can aid selection of an optimal signal processing strategy. In this paper we present a method for estimation of direct-to-reverberant energy ratio from partial room responses to an impulsive sound at one receiver location, and of source distance, average absorption, reverberation time, and room volume as intermediate results.

Bimodal stimulus presentation and expanded auditory bandwidth improve older adults' speech perception.

Objective: A pair of experiments investigated the hypothesis that bimodal (auditory-visual) speech presentation and expanded auditory bandwidth would improve speech intelligibility and increase working memory performance for older adults by reducing the cognitive effort needed for speech perception. Background: Although telephone communication is important for helping older adults maintain social engagement, age-related sensory and working memory limits may make telephone conversations difficult. Method: Older adults with either age-normal hearing or mild-to-moderate sensorineural hearing loss performed a running memory task. Participants heard word strings of unpredictable length and at the end of each string were required to repeat back the final three words. Words were presented monaurally in telephone bandwidth (300 Hz to 3300 Hz) or expanded bandwidth (50 Hz to 7500 Hz), in quiet (65 dBZ SPL), or in white noise (65 dBZ SPL with noise at 60 dBZ SPL), with or without a visual display of the talker. Results: In quiet listening conditions, bimodal presentation increased the number of words correctly reported per trial but only for listeners with hearing loss and with high lipreading proficiency. Stimulus bandwidth did not affect performance. In noise, bimodal presentation and expanded bandwidth improved performance for all participant groups but did so by improving speech intelligibility, not by improving working memory. Conclusion: Expanded bandwidth and bimodal presentation can improve speech perceptibility in difficult listening conditions but may not always improve working memory performance. Application: Results can inform the design of telephone features to improve ease of communication for older adults.

A minimum‐variance frequency‐domain algorithm for binaural hearing aid processing

A new algorithm has been developed that allows optimal filtering methods to be applied individually to different narrow frequency bands. Using this technique it is possible to process binaural signals in a manner which dramatically reduces the amplitude of signals originating away from a desired receive direction. The algorithm was tested on artificially combined anechoic and reverberant signals with varying numbers of interfering sound sources. In addition to being computationally efficient, the algorithm was able to produce output which had a consistently positive intelligibility weighted SNR gain [Link and Buckley, J. Acoust. Soc. Am. 91, 1662 (1992)], and in which the desired talker was noticeably easier to understand. Results of a paired‐comparison listening test confirmed these results, and allowed an estimate to be made of the algorithm parameters which provided the best speech intelligibility for the output. These parameters included the window length, length of a filtered block, and amount of dat...

Seeing facial motion affects auditory processing in noise

Speech perception, especially in noise, may be maximized if the perceiver observes the naturally occurring visual-plus-auditory cues inherent in the production of spoken language. Evidence is conflicting, however, about which aspects of visual information mediate enhanced speech perception in noise. For this reason, we investigated the relative contributions of audibility and the type of visual cue in three experiments in young adults with normal hearing and vision. Relative to static visual cues, access to the talker’s phonetic gestures in speech production, especially in noise, was associated with (a) faster response times and sensitivity for speech understanding in noise, and (b) shorter latencies and reduced amplitudes of auditory N1 event-related potentials. Dynamic chewing facial motion also decreased the N1 latency, but only meaningful linguistic motions reduced the N1 amplitude. The hypothesis that auditory–visual facilitation is distinct to properties of natural, dynamic speech gestures was partially supported.

Human performance in a multisource environment with a frequency‐banded minimum‐variance beamforming algorithm

The effectiveness of a newly developed signal‐processing algorithm to extract speech in the presence of multiple interferers was evaluated through intelligibility ratings from normal‐ and hearing‐impaired listeners. Sentences were presented with competing speech and multitalker babble (MTB) at four different SNRs to groups of individuals with sensorineural hearing loss and normal hearing. The sentences and competition had been processed through a real‐time, frequency‐banded minimum‐variance beamforming (FBMVB) algorithm in multitalker environments (Lockwood et al., 1999). Competing messages and MTB were simulated at +22 and +45 azimuths relative to the target source. Each target sentence with competing messages was submitted to the FBMVB algorithm to yield a set of 24 ‘‘processed’’ and 24 ‘‘unprocessed’’ sentences per SNR condition. Listeners rated the intelligibility (0%–100%) of the target sentences and repeated the last word of each sentence. Results from the ratings and the word recognition scores sho...