Joseph L. Hall

Stereophonic acoustic echo cancellation-an overview of the fundamental problem

As teleconferencing systems evolve to an ever more lifelike and transparent audio/video medium, it will be necessary to incorporate multichannel audio, which at a minimum involves two channels, i.e., stereophonic sound. However, before full-duplex stereophonic teleconferencing can be deployed, the acoustic echo cancellation (AEC) problem must be solved in this regime. This paper draws attention to a fundamental problem of multichannel AEC that concerns the nonunique nature of the estimated receiving room impulse responses. Various potential solutions are discussed but are shown to be not entirely satisfactory.<<ETX>>

Model for mechanical to neural transduction in the auditory receptor

We describe a model for transduction of displacement of the basilar membrane to activity of auditory nerve fibers. The model is physiologically oriented. Quanta (“vesicles”) are added to a population inside an urn (hair cell) at a fixed average rate. They are removed from the population with a probability proportional to the number of quanta in the population and related in a simple manner to the instantaneous amplitude of the input signal (displacement). The removal of a quantum results in an event (action potential) with a probability related to time elapsed since the preceding event (refractory period). Apart from refractoriness, the model is completely specified by three parameters which determine spontaneous and maximum firing rates and the time constant of recovery after intense stimulation. The model is more compatible with observed activity of the auditory nerve than threshold‐crossing models. An important feature of auditory‐nerve data reproduced by the model is the level normalization observed at moderate to high stimulus intensities. The model reproduces features of period, PST, and interval histograms in response to pure and complex tones, tone bursts, and noise. We present results both from analysis and from computer simulation of the model.

Investigation of several types of nonlinearities for use in stereo acoustic echo cancellation

In this paper, we investigate several types of nonlinearities used for the unique identification of receiving room impulse responses in stereo acoustic echo cancellation. The effectiveness is quantified by the mutual coherence of the transformed signals. The perceptual degradation is studied by psycho-acoustic experiments in terms of subjective quality and localization accuracy in the medial plane. The results indicate that, of the several nonlinearities considered, ideal half-wave rectification appears to be the best choice for speech. For music, the nonlinearity parameter of the ideal rectifier must be readjusted. The smoothed rectifier does not require this readjustment, but is a little more difficult to implement.

Stereophonic acoustic echo cancellation using nonlinear transformations and comb filtering

Stereophonic sound becomes more and more important in a growing number of applications (such as teleconferencing, multimedia workstations, televideo gaming, etc.) where spatial realism is demanded. Such hands-free systems need stereophonic acoustic echo cancelers (AECs) to reduce echos that result from coupling between loudspeakers and microphones in full-duplex communication. We propose a new stereo AEC based on two experimental observations: (a) the stereo effect is due mostly to sound energy below about 1 kHz and (b) comb filtering above 1 kHz does not degrade auditory localization. The principle of the proposed structure is to use one stereo AEC at low frequencies (e.g. below 1 kHz) with nonlinear transformations on the input signals and another stereo AEC at higher frequencies (e.g. above 1 kHz) with complementary comb filters on the input signals.

Asymmetry of masking revisited: Generalization of masker and probe bandwidth

A band of noise masked by a tone is audible at a much lower intensity increment between masker alone and masker plus just-detectable probe than is a tone masked by a band of noise. To better understand the mechanisms responsible for this asymmetry of masking, we measured thresholds of probe signals ranging in bandwidth from 0 to 256 Hz in the presence of masking signals also ranging in bandwidth from 0 to 256 Hz. Masker and probe center frequencies were 1 kHz in all cases. Two versions of the experiment are reported: (1) masker intensity constant at 70 dB SPL and (2) masker intensity roved over the range 65–75 dB SPL. The results confirm and extend previous results. For a fixed masker bandwidth, threshold intensity increments are essentially constant, and accounted for by a model based on long-term average energy, so long as probe bandwidth does not exceed masker bandwidth. If probe bandwidth exceeds masker bandwidth, threshold intensity increments decrease sharply for all masker bandwidths tested. Roved-intensity results are consistent with the hypothesis that detection is based on long-term average energy for probe bandwidth equal to or less than masker bandwidth, but that for probe bandwidth greater than masker bandwidth other cues are utilized. In addition, we investigated the possible contribution of off-frequency listening by developing and presenting results from a frequency-selective model of peripheral masking. Model results argue against detection based on spectral cues associated with off-frequency listening for signal bandwidths less than a critical band. The conclusion is that detection must utilize the temporal structure of the signal for probe bandwidth greater than masker bandwidth.

Application of multidimensional scaling to subjective evaluation of coded speech

We present results from a pilot study directed at developing an anchorable subjective speech quality test. The test uses multidimensional scaling techniques to obtain quantitative information about the perceptual attributes of speech. In the first phase of the study, subjects ranked perceptual distances between samples of speech produced by two different talkers, one male and one female, processed by a variety of codecs. The resulting distance matrices were processed to obtain, for each talker, a stimulus space for the various speech samples. This stimulus space has the properties that distances between stimuli in this space correspond to perceptual distances between stimuli and that the dimensions of this space correspond to attributes used by the subjects in determining perceptual distances. Mean opinion scores (MOS) scores obtained in an earlier study were found to be highly correlated with position in the stimulus space, and the three dimensions of the stimulus space were found to have identifiable physical and perceptual correlates. In the second phase of the study, we developed techniques for fitting speech generated by a new codec under investigation into a previously established stimulus space. The user is provided with a collection of speech samples and with the stimulus space for these speech samples as determined by a large-scale listening test. The user then carries out a much smaller listening test to determine the position of the new stimulus in the previously established stimulus space. This system is anchorable, so that different versions of a codec under development can be compared directly, and it provides more detailed information than the single number provided by MOS testing. We suggest that this information could be used to advantage in algorithm development and in development of objective measures of speech quality.

Synthesized stereo combined with acoustic echo cancellation for desktop conferencing

One promising application in communications is desktop conferencing, which can involve several participants over a widely distributed area. Synthesized stereophonic sound will enable a listener to spatially separate one remote talker from another and thereby improve understanding. In such a scenario, we assume we are located in a hands-free environment where the composite acoustic signal is presented over loudspeakers, thus requiring acoustic echo cancellation. In this paper, we explain some of the methods that can be used to synthesize stereo sound and how such methods can be combined efficiently with stereo acoustic echo cancellation in the face of several difficult problems.One promising application in modern communications is desktop conferencing, which can involve several participants over a widely distributed area. Single-channel monophonic sound certainly is not sufficient for a listener to spatially separate one remote talker from another. Moreover, it is very difficult to understand two (or more) remote speakers talking at the same time. Synthesized stereophonic sound will help considerably in solving these problems. In such a scenario, we assume we are located in a hands-free environment, where the composite acoustic signal is presented over loudspeakers, thus requiring acoustic echo cancellation. In this paper, we explain some of the methods that can be used to synthesize stereo sound and how such methods can be combined efficiently with stereo acoustic echo cancellation in the face of several difficult problems.

Acoustic echo cancellation for stereophonic teleconferencing

In long‐distance telephony, echoes arise due to impedance mismatches at various points in the telephone circuit. Adaptive line echo cancelers have been used successfully for over a decade to combat this problem. Echoes also arise in teleconferencing, due to acoustic coupling between microphone and loudspeaker in each conference room. This problem is similar to the line echo problem; however, the echo paths are much longer and much more variable in this case. In this paper a further complication that arises if stereophonic transmission is used for teleconferencing is discussed: There is an inherent nonuniqueness in estimating the echo paths. It appears that the only way to resolve this nonuniqueness is by somehow decorrelating the signals in the two stereo channels. Several methods of decorrelation are discussed and how they affect adaptive echo canceller performance as well as stereophonic perception is shown.

Peak detection for auditory sound discrimination

Auditory detection of envelope maxima in temporal responses of cochlear frequency‐analyzing filters has been hypothesized to account for phase effects in psychophysical discrimination [J. L. Goldstein, 458–479 (1967)]. Re‐examination of this hypothesis in the context of asymmetry of masking [R. Hellman, Percept. Psychophys. 11, 241–246 (1972)] reveals that it also provides an adequate explanation for this phenomenon. Peak discrimination between a tone and tone masker plus narrow‐band‐noise probe is more sensitive to probe energy than is the inverse discrimination between noise and noise masker plus tone probe, in agreement with psychophysics. Simulations of this model indicate that asymmetry of masking is a function of the product of noise bandwidth and temporal duration. Psychophysical experiments on masking asymmetry were performed with both masker and probe bandwidth ranging from pure tone to supracritical band. The experimental design included both fixed and roving levels, with random phases fixed thr...

Temporal factors in auditory peak detection of modulation of tones

Auditory detection of tone modulation has been modeled as envelope‐peak detection of the cochlear filterbank responses [J. L. Goldstein, J. Acoust. Soc. Am. 41, 458–479 (1967)]. To study the ability of this model to quantify temporal smoothing in detection, new psychophysical experiments using a 4IFC paradigm were conducted, with the authors as subjects. AM or quasi‐FM sinusoidally modulated tones were discriminated from the carrier tone. Only one should was modulated among the four pushed sounds in the paradigm. The first series of exploratory experiments (1994) focused on low modulation frequencies (4–16 Hz). A second series of systematic experiments (1996) with a wide range of modulation frequencies allowed estimation of detector smoothing at carrier frequencies of 0.25, 1, and 4 kHz. Despite differences between subjects in estimated spectral filtering, modulation thresholds for both subjects gave similar estimates of smoothing at each carrier frequency. At modulation frequencies above 20 Hz, the smoot...