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检索条件"主题词=Linear predictive coding"

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Speaker verification using temporal decorrelation post-processing

Speaker verification using temporal decorrelation post-proce...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： L.P. Netsch G.R. Doddington Computer Science Laboratory Texas Instruments Inc. Dallas TX USA SRI International Inc. Menlo Park CA USA

A text-dependent method of speaker verification processing which utilizes the statistical correlation between measured features of speech across whole words is described. The correlation is used in a linear discriminant analysis to define uncorrelated world-level features as a metric. Initial results indicate that this method can significantly reduce the amount of storage necessary for speaker-specific speech information. Furthermore, this method provides promise of improved verification performance compared to methods based on hidden Markov model (HMM) state level observation metrics. Since the linear discriminant analysis yields features which are decorrelated over entire words, this method should be more robust to signal distortions which are consistent over the entire utterance.< >

关键词： Decorrelation Hidden Markov models Speech processing Vectors Statistics Telephone sets Feature extraction linear predictive coding Filter bank Nonlinear filters

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Secure communication between tactical and strategic networks

Secure communication between tactical and strategic networks

引用

MILCOM, Military Communications Conference

作者： J. Bertrand D. Caprioni C. Haight AeroSpace/Communications Division ITT Clifton NJ USA ITT Aerosp./Commun. Div. Clifton NJ USA

The STU-III program has been very successful in providing secure, high quality communications. The STU-III program, however, has so far been restricted to strategic networks. Tactical networks use devices such as SINCGARS, VINSON, and MINTERM (tactical terminal) which can not interoperate with the STU-III network. The paper discusses the modifications to MINTERM that provide it with a STU-III-compatible mode. It is shown that most of the changes can be made in software and that the modified MINTERM can interoperate with the STU-III network over a large variety of media.

关键词： Modems Switches Digital communication Telephone sets linear predictive coding Communication switching Tin Hafnium

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A combined quantization-interpolation scheme for very low bit rate coding of speech LSP parameters

A combined quantization-interpolation scheme for very low bi...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： J.M. Lopez-Soler N. Farvardin Department Electrónica y Tecnología de Computadores Facultad de Ciencias Universidad de Granada Granada Spain

A combined quantization-interpolation of speech line spectrum pair (LSP) parameters is proposed. To utilize the linear dependency between successive LSP frames, the proposed algorithm attempts to locate the frames where there is a significant spectral change; these frames are encoded by vector quantization. The remaining frames are reconstructed by linear interpolation between the vector-quantized frames. A two-pass algorithm is used to couple the quantization and interpolation operations. Simulation results for the performance of the combined quantization-interpolation approach indicate that, at rates below 350 bits/s (for coding the spectral parameters), the proposed scheme outperforms the multiframe coding scheme of D. P. Kemp et al. (ICASSP-91, p.609-12). The effect of maximum delay on the overall performance is studied, and the performance of the proposed system over noisy channels is examined.< >

关键词： Bit rate Speech coding Interpolation Rate-distortion linear predictive coding Encoding Educational institutions Vector quantization Delay effects Vocoders

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Multi-level clustering of acoustic features for phoneme recognition based on mutual information

Multi-level clustering of acoustic features for phoneme reco...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： K. Shirai N. Aoki N. Hosaka Department of Electrical Engineering Waseda University Shinjuku Tokyo Japan

An optimal method for organizing acoustic features to recognize phonemes in continuous speech is described. Each level of acoustic features, including power and its variational pattern, and the linear predictive coding Mel-cepstrum and its pattern of temporal change, is clustered hierarchically on the basis of the mutual information between the acoustic feature vector and phoneme labels assigned for the speech wave. Multilevel clustering is used to discriminate phonemes by detecting the most reliable features in the context and by using the effective combination of acoustic characteristics. Phoneme recognition for each frame is discussed. The conditional entropy is evaluated for the phoneme labels of the frame, given the various acoustic features for the neighboring frames. Phoneme discrimination can be performed effectively using the conditional entropy. In the preliminary test the phoneme recognition rate was 81.6%, and the vowel recognition rate was 92.4% in the frame level. In a completely talker-independent experiment the recognition rates were 76.8% and 89.7%, respectively.< >

关键词： Speech recognition Acoustic waves Entropy Organizing linear predictive coding Mutual information Vectors Speech coding Acoustic signal detection Acoustic testing

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A vector quantizer incorporating both LPC shape and energy

A vector quantizer incorporating both LPC shape and energy

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： L. Rabiner M. Sondhi S. Levinson Bell Laboratories Murray Hill NJ USA

The theory of vector quantization (VQ) of linear predictive coding (LPC) coefficients has established a wide variety of techniques for quantizing LPC spectral shape to minimize overall spectral distortion. Such vector quantizers have been widely used in the areas of speech coding and speech recognition. The conventional vector quantizer utilizes only spectral shape information and essentially disregards the energy or gain term associated with the optimal LPC fit to the signal being modelled. In this paper we present a method of incorporating LPC spectral shape and energy into the codebook entries of the vector quantizer. To do this we postulate a distortion measure for comparing two LPC vectors which uses a weighted sum of an LPC shape distortion and a log energy distortion. Based on this combined distortion measure we have designed and studied vector quantizers of several sizes for use in isolated word speech recognition experiments. We have found that a fairly significant correlation exists between LPC shape and signal energy; hence a combined LPC shape plus energy vector quantizer with a given distortion requires far fewer codebook entries than one in which LPC shape and energy are quantized separately. Based on isolated word recognition tests on both a 10-digit and a 129 word airlines vocabulary, we have found improvements in recognition accuracy by using the VQ with both LPC shape and energy over that obtained using a VQ with LPC shape alone.

关键词： linear predictive coding Distortion measurement Spectral shape Speech recognition Shape measurement Vector quantization Speech coding Energy measurement Size measurement Testing

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Improvement of speech enhancement techniques for robust speaker identification in noise

Improvement of speech enhancement techniques for robust spea...

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International Conference on Computer and Information Technology (ICCIT)

作者： Md. Rabiul Islam Md. Fayzur Rahman Muhammad Abdul Goffar Khan Department of Computer Science & Engineering Rajshahi University of Engineering & Technology Rajshahi Bangladesh Department of Electrical & Electronic Engineering Rajshahi University of Engineering & Technology Rajshahi Bangladesh Department of Electrical & Electronic Engineering Rajshahi University of Engineering & Technology Bangladesh

This paper presents an approach of speech enhancement techniques to improve the performance of the robust speaker identification under noisy environments. Start-end points detection, silence part removal, frame segmentation and windowing technique have been used to pre-process and Wiener filter has been used to remove the silence parts from the speech utterances. To extract the features from the speech various speech parameterization techniques that is LPC, LPCC, RCC, MFCC, ¿MFCC and ¿¿MFCC have been simulated. Finally, to measure the performance of the proposed speech enhancement techniques, genetic algorithm has been used as a classifier for the noise robust automated speaker identification system and various experiments have performed on genetic algorithm to select the optimum parameters. According to the NOIZEOUS speech database, the highest identification rate of 70.31 [%] for text-dependent and of 61.26 [%] for text-independent speaker identification system have been achieved.

关键词： Speech enhancement Noise robustness Working environment noise Genetic algorithms Wiener filter Feature extraction linear predictive coding Mel frequency cepstral coefficient Noise measurement Performance evaluation

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Determination of the LPC coefficients using neural networks

Determination of the LPC coefficients using neural networks

引用

International Conference on Neural Networks

作者： I.I. Ibrahim M.I. El-Adawy Electronics and Communications Department Faculty of Engineering and Technology Cairo Egypt Helwan University Cairo EG Dept. of Electron. & Commun. Helwan Univ. Cairo Egypt

A new technique for the determination of the LPC coefficients is introduced. In this method the LPC coefficients are shown to be the weighting factors of a simple neuron. This makes this technique suitable for a lot of applications that uses LPC as a signal processing tool like speech processing and production. The error introduced by this method is comparable to the error obtained by the conventional methods like the autocorrelation method.

关键词： linear predictive coding Neural networks Differential equations Neurons Autocorrelation Signal processing algorithms Testing Vectors Nonlinear equations Speech processing

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Effects of hybrid nonlinearity on a full-duplex telephone network with vocoder

Effects of hybrid nonlinearity on a full-duplex telephone ne...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： D. Mansour A. Gray Department of Electrical & Computer Engineering University of California Santa Barbara USA Signal Technology Inc. University of California Santa Barbara USA

In most applications of adaptive filtering algorithms to the telephone echo cancellation problem, a linear model is assumed for the hybrid. In this paper, a nonlinear model based on a Volterra series representation is adopted. By using a new and efficient nonlinear frequency domain adaptive algorithm, a nonlinear model for a hybrid in an actual connection has been identified. The nonlinear model is found to be 8 db more accurate than the linear model. The problem of automatic gain control in the design of a full-duplex telephone network with LPC vocoder in the loop, and its interaction with the echo canceller and the hybrid nonlinearity is also discussed.

关键词： Telephony Vocoders Frequency Echo cancellers Convergence Tires Costs Gain control linear predictive coding Data communication

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Improved phonetically-segmented vector excitation coding at 3.4 kb/s

Improved phonetically-segmented vector excitation coding at ...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： S. Wang A. Gersho Center for Information Processing Research Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

Several major modifications to the phonetically segmented vector excitation coding (PS-VXC) coder by the authors (1989, 1990) reported previously have resulted in enhanced speech quality while reducing the delay, complexity, and bit rate. Speech is segmented into variable-length phonetic classes and a VXC coding module is tailored to each class. coding techniques include adaptive linear predictive coding (LPC) analysis and interpolation, two-stage excitation coding of onsets, comb filtering, modified perceptual weighting, and pitch contour smoothing. The improved PS-VXC coder operates at a peak rate of 3.4 kb/s with an average rate of 3.0 kb/s and has a subjective performance closely matching that of the 4.8 kb/s DoD CELP coder.< >

关键词： Speech coding linear predictive coding Vectors Speech enhancement Delay Bit rate Interpolation Adaptive filters Nonlinear filters Filtering

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Baseband LPC coders for speech transmission over 9.6 kb/s noisy channels

Baseband LPC coders for speech transmission over 9.6 kb/s no...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： R. Viswanathan A. Higgins W. Russell J. Makhoul Bolt Beranek and Newman Inc. Cambridge Mass. Bolt Beranek and Newman 슠Inc. Cambridge MA USA

This paper presents the results of our investigation of the various aspects of baseband LPC coders with the goal of maximizing the speech quality at a transmission bit-rate of 9.6 kb/s and for channel bit-error rates of up to 1%. Important among these aspects are: baseband width, coding of baseband, high-frequency regeneration, and error protection of important transmission parameters. The paper discusses these and other issues, presents the results of speech-quality tests conducted during the various stages of optimization, and describes the details of the optimized speech coder.

关键词： Baseband linear predictive coding Speech Signal processing Protection Bandwidth Data mining Phase change materials Fasteners Testing

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