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Journal ArticleDOI

Statistical analysis of the autoregressive modeling of reverberant speech

TLDR
Analytical results from statistical room acoustics are utilizes to analyze the AR modeling of speech under reverberant conditions and it is demonstrated that at each individual source-microphone position (without spatial expectation), the M-channel AR coefficients provide the best approximation to the clean speech coefficients when microphones are closely spaced.
Abstract
Hands-free speech input is required in many modern telecommunication applications that employ autoregressive (AR) techniques such as linear predictive coding. When the hands-free input is obtained in enclosed reverberant spaces such as typical office rooms, the speech signal is distorted by the room transfer function. This paper utilizes theoretical results from statistical room acoustics to analyze the AR modeling of speech under these reverberant conditions. Three cases are considered: (i) AR coefficients calculated from a single observation; (ii) AR coefficients calculated jointly from an M-channel observation (M > 1); and (iii) AR coefficients calculated from the output of a delay-and sum beamformer. The statistical analysis, with supporting simulations, shows that the spatial expectation of the AR coefficients for cases (i) and (ii) are approximately equal to those from the original speech, while for case (iii) there is a discrepancy due to spatial correlation between the microphones which can be significant. It is subsequently demonstrated that at each individual source-microphone position (without spatial expectation), the M-channel AR coefficients from case (ii) provide the best approximation to the clean speech coefficients when microphones are closely spaced (<0.3m).

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Citations
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Single- and multi-microphone speech dereverberation using spectral enhancement

TL;DR: Novel single- and multimicrophone speech dereverberation algorithms are developed that aim at the suppression of late reverberation, i.e., signal processing techniques to reduce the detrimental effects of reflections.
Journal ArticleDOI

Temporal Dynamics for Blind Measurement of Room Acoustical Parameters

TL;DR: Experiments suggest that estimators of subjective perception of spectral coloration, reverberant tail effect, and overall speech quality can be obtained with an adaptive speech-to-reverberation modulation energy ratio measure.
Journal ArticleDOI

Regularization for Partial Multichannel Equalization for Speech Dereverberation

TL;DR: A partial multichannel equalization technique based on MINT (P-MINT) is proposed which aims to shorten the RIR and an automatic non-intrusive procedure for determining the regularization parameter based on the L-curve is introduced.
Proceedings ArticleDOI

Spatiotemporal Averagingmethod for Enhancement of Reverberant Speech

TL;DR: A reverberant speech enhancement algorithm which, operating on the linear prediction residual of spatially averaged multi- microphone observations, utilizes temporal averaging of neighbouring larynx cycles to design an equalization filter to dereverberate both voiced and unvoiced speech.
Journal ArticleDOI

Speech enhancement for robust automatic speech recognition: Evaluation using a baseline system and instrumental measures

TL;DR: Results of an evaluation of different speech enhancement pipelines using a state-of-the-art ASR system for a wide range of reverberation and noise conditions indicate the deleterious effect of both noise and reverberation.
References
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TL;DR: This paper gives an exposition of linear prediction in the analysis of discrete signals as a linear combination of its past values and present and past values of a hypothetical input to a system whose output is the given signal.
Journal ArticleDOI

Beamforming: a versatile approach to spatial filtering

TL;DR: An overview of beamforming from a signal-processing perspective is provided, with an emphasis on recent research.
Journal ArticleDOI

Image method for efficiently simulating small‐room acoustics

TL;DR: The theoretical and practical use of image techniques for simulating the impulse response between two points in a small rectangular room, when convolved with any desired input signal, simulates room reverberation of the input signal.
Book

Discrete-Time Processing of Speech Signals

TL;DR: The preface to the IEEE Edition explains the background to speech production, coding, and quality assessment and introduces the Hidden Markov Model, the Artificial Neural Network, and Speech Enhancement.