This paper presents an analysis of factors affecting system performance in the ASpIRE (Automatic Speech recognition In Reverberant Environments) challenge. In particular, overall word error rate (WER) of the solver systems is analyzed as a function of room, distance between talker and microphone, and microphone type. We also analyze speech activity detection performance of the solver systems and investigate its relationship to WER. The primary goal of the paper is to provide insight into the factors affecting system performance in the ASpIRE evaluation set across many systems given annotations and metadata that are not available to the solvers. This analysis will inform the design of future challenges and provide insight into the efficacy of current solutions addressing noisy reverberant speech in mismatched conditions.

Previous studies have shown that, in addition to being speaker-dependent yet context-independent, lower vocal tract acoustics significantly impact the speech spectrum at mid-tohigh frequencies (e.g 3-6kHz). The present work automatically estimates spectral features that exhibit acoustic properties of the lower vocal tract. Specifically aiming to capture the cyclicity property of the epilarynx tube, a novel multi-resolution approach to spectral analyses is presented that exploits significant differences between the closed and open phases of a glottal cycle. A prominent null linked to the piriform fossa is also estimated. Examples of the feature estimation on natural speech of the VOICES multi-speaker corpus illustrate that a salient spectral pattern indeed emerges between 3-6kHz across all speakers. Moreover, the observed pattern is consistent with that canonically shown for the lower vocal tract in previous works. Additionally, an instance of a speaker’s formant (i.e. spectral peak around 3kHz that has been well-established as a characteristic of voice projection) is quantified here for the VOICES template speaker in relation to epilarynx acoustics. The corresponding peak is shown to be double the power on average compared to the other speakers (20 vs 10 dB).

We introduce a framework for speech enhancement based on convolutive non-negative matrix factorization that leverages available speech data to enhance arbitrary noisy utterances with no a priori knowledge of the speakers or noise types present. Previous approaches have shown the utility of a sparse recon-struction of the speech-only components of an observed noisy utterance. We demonstrate that an underlying speech represen-tation which, in addition to applying sparsity, also adapts to the noisy acoustics improves overall enhancement quality. The proposed system performs comparably to a traditional Wiener filtering approach, and the results suggest that the proposed framework is most useful in moderate- to low-SNR scenarios. Index Terms: speech enhancement, convolutive non-negative matrix factorization, basis adaptation, sparsity

1 A hypothesis in characterizing human depression is that change in the brain‟s basal ganglia results in a decline of motor coordination [6][8][14]. Such a neuro-physiological change may therefore affect laryngeal control and dynamics. Under this hypothesis, toward the goal of objective monitoring of depression severity, we investigate vocal-source biomarkers for depression; specifically, source features that may relate to precision in motor control, including vocal-fold shimmer and jitter, degree of aspiration, fundamental frequency dynamics, and frequency-dependence of variability and velocity of energy. We use a 35-subject database collected by Mundt et al. [1] in which subjects were treated over a six-week period, and investigate correlation of our features with clinical (HAMD), as well as self-reported (QIDS) Total subject assessment scores. To explicitly address the motor aspect of depression, we compute correlations with the Psychomotor Retardation component of clinical and self-reported Total assessments. For our longitudinal database, most correlations point to statistical relationships of our vocal-source biomarkers with psychomotor activity, as well as with depression severity.

The accurate estimation of turbulence noise affects many areas of speech processing including separate modification of the noise component, analysis of degree of speech aspiration for treating pathological voice, the automatic labeling of speech voicing, as well as speaker characterization and recognition. Previous work in the literature has provided methods by which such a high-quality noise component may be estimated in near-periodic speech, but it is known that these methods tend to leak aperiodic phonation (with even slight deviations from periodicity) into the noise-component estimate. In this paper, we improve upon existing algorithms in conditions of aperiodicity by introducing a time-warping based approach to speech noise-component estimation, demonstrating the results on both natural and synthetic speech examples.

Regions of nonmodal phonation, exhibiting deviations from uniform glottal-pulse periods and amplitudes, occur often and convey information about speaker- and linguistic-dependent factors. Such waveforms pose challenges for speech modeling, analysis/synthesis, and processing. In this paper, we investigate the representation of nonmodal pulse trains as a sum of harmonically-related sinewaves with time-varying amplitudes, phases, and frequencies. We show that a sinewave representation of any impulsive signal is not unique and also the converse, i.e., frame-based measurements of the underlying sinewave representation can yield different impulse trains. Finally, we argue how this ambiguity may explain addition, deletion, and movement of pulses in sinewave synthesis and a specific illustrative example of time-scale modification of a nonmodal case of diplophonia. Index Terms: sinewave analysis, sinewave synthesis, nonmodal phonation

Abstract 1 Of increasing importance in the civilian and military population is the recognition of Major Depressive Disorder at its earliest stages and intervention before the onset of severe symptoms. Toward the goal of more effective monitoring of depression severity, we investigate automatic classifiers of depression state, that have the important property of mitigating nuisances due to data variability, such as speaker and channel effects, unrelated to levels of depression. To assess our measures, we use a 35-speaker free-response speech database of subjects treated for depression over a six-week duration, along with standard clinical HAMD depression ratings. Preliminary experiments indicate that by mitigating nuisances, thus focusing on depression severity as a class, we can significantly improve classification accuracy over baseline Gaussian-mixture-model-based classifiers. Index Terms: major depressive disorder, Gaussian-mixture models, joint factor analysis, nuisance mitigation

The global and fine dynamic components of a pitch contour in voice production, as in the speaking and singing voice, are important for both the meaning and character of an utterance. In speech, for example, slow pitch inflections, rapid pitch accents, and irregular regions all comprise the pitch contour. In applications where all components of a pitch contour are stretched or compressed in the same way, as for example in time-scale modification, an unnatural scaled contour may result. In this paper, we develop a framework for scaling pitch contours, motivated by the goal of maintaining naturalness in time-scale modification of voice. Specifically, we develop a multi-band algorithm to independently modify the slow trajectory and fast perturbation components of a contour for a more natural synthesis, and we present examples where pitch contours representative of speaking and singing voice are lengthened. In the speaking voice, the frequency content of flutter or irregularity is maintained, while slow pitch inflection is simply stretched or compressed. In the singing voice, rapid vibrato is preserved while slower note-to-note variation is scaled as desired.

Regions of nonmodal phonation, which exhibit deviations from uniform glottal-pulse periods and amplitudes, occur often in speech and convey information about linguistic content, speaker identity, and vocal health. Some aspects of these deviations are random, including small perturbations, known as jitter and shimmer, as well as more significant aperiodicities. Other aspects are deterministic, including repeating patterns of fluctuations such as diplophonia and triplophonia. These deviations are often the source of misinterpretation of the spectrum. In this paper, we introduce a general signal-processing framework for interpreting the effects of both stochastic and deterministic aspects of nonmodality on the short-time spectrum. As an example, we show that the spectrum is sensitive to even small perturbations in the timing and amplitudes of glottal pulses. In addition, we illustrate important characteristics that can arise in the spectrum, including apparent shifting of the harmonics and the appearance of multiple pitches. For stochastic perturbations, we arrive at a formulation of the power-spectral density as the sum of a low-pass line spectrum and a high-pass noise floor. Our findings are relevant to a number of speech-processing areas including linear-prediction analysis, sinusoidal analysis-synthesis, spectrally derived features, and the analysis of disordered voices.

Nonmodal phonation occurs when glottal pulses exhibit nonuniform pulse-to-pulse characteristics such as irregular spacings, amplitudes, and/or shapes. The analysis of regions of such nonmodality has application to automatic speech, speaker, language, and dialect recognition. In this paper, we examine the usefulness of a technique called minimum-entropy deconvolution, or MED [1], for the analysis of pulse events in nonmodal speech. Our study presents evidence for both natural and synthetic speech that MED decomposes nonmodal phonation into a series of sharp pulses and a set of mixedphase impulse responses. We show that the estimated impulse responses are quantitatively similar to those in our synthesis model. A hybrid method incorporating aspects of both MED and linear prediction is also introduced. We show preliminary evidence that the hybrid method has benefit over MED alone for composite impulse-response estimation by being more robust to short-time windowing effects as well as a speech aspiration noise component. Index Terms: inverse filtering, nonmodal speech, glottal pulse, minimum entropy

A dysphonia, or disorder of the mechanisms of phonation in the larynx, can create time-varying amplitude fluctuations in the voice. A model for band-dependent analysis of this amplitude modulation (AM) phenomenon in dysphonic speech is developed from a traditional communications engineering perspective. This perspective challenges current dysphonia analysis methods that analyze AM in the time-domain signal. An automatic dysphonia recognition system is designed to exploit AM in voice using a biologically inspired model of the inferior colliculus. This system, built upon a Gaussian-mixture-model (GMM) classification backend, recognizes the presence of dysphonia in the voice signal. Recognition experiments using data obtained from the Kay elemetrics voice disorders database suggest that the system provides complementary information to state-of-the-art mel-cepstral features. We present dysphonia recognition as an approach to developing features that capture glottal source differences in normal speech.