Introduction to Digital Speech Processing

DOI10.1561/2000000001MaRDI QIDQ5443391zbMATH OpenOpenAlexFDO

Authors Lawrence R. Rabiner, Ronald W. Schafer

Publication date 20 February 2008

Published in Foundations and Trends® in Signal Processing (Search for Journal in Brave)

Full work available at URL https://doi.org/10.1561/2000000001

zbMATH Keywords

speech recognitionspeech synthesisspeech coding

Mathematics Subject Classification ID

Pattern recognition, speech recognition (68T10) Signal theory (characterization, reconstruction, filtering, etc.) (94A12) Introductory exposition (textbooks, tutorial papers, etc.) pertaining to information and communication theory (94-01)

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Cited in

(24)

• A mathematical motivation for complex-valued convolutional networks
• Linear Prediction of Speech
• Measuring statistical dependences in a time series
• Hints about some baseful but indispensable elements in speech recognition and reconstruction
• Advances in non-linear modeling for speech processing
• A tree-weighting approach to sequential decision problems with multiplicative loss
• Detection of the glottal closure instants using empirical mode decomposition
• Survey article: continued fractions associated with Wiener-Levinson filters, frequency analysis, moment theory and polynomials orthogonal on the unit circle
• An inverse problem to determine the shape of a human vocal tract
• On generalized predictive compression of continuous-time signal samples
• Digital speech processing using Matlab
• Optimal reconstruction of ARMA signals with decimated samples under corrupting noise by use of multirate Kalman filter
• Hybrid modeling design patterns
• On the method of finding frequencies with large amplitudes
• Minimum-phase parts of zero-phase sequences
• Quality analog scramblers using frequency-response masking filter banks
• Asymptotics for Szegö polynomial zeros
• Moment problems and low rank Toeplitz approximations
• Principles of speech coding.
• Determining the shape of a human vocal tract from pressure measurements at the lips
• CID: an efficient complexity-invariant distance for time series
• System identification techniques for adaptive signal processing
• Statistically optimal null filter based on instantaneous matched processing
• Mathematical modeling and signal processing in speech and hearing sciences

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