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检索条件"主题词=linear predictive coding"

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On nonlinear utilization of intervector dependency in vector quantization

On nonlinear utilization of intervector dependency in vector...

引用

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

作者： M. Skoglund J. Skoglund Department of Signals Sensors and Systems Royal Institute of Technology Stockholm Sweden Department of Information Theory Chalmers University of Technology Goteborg Sweden

This paper presents an approach to speech vector quantization of sources exhibiting intervector dependency. We present the optimal decoder based on a collection of received indices. We also present the optimal encoder for such decoding. The optimal decoder can be implemented as a table look-up decoder, however the size of the decoder codebook grows very fast with the size of the collection of utilized indices. This leads us to introduce a method for storing an approximation to the set of optimal decoder vectors, based on linear mapping of a block code vector quantization. In this approach a heavily reduced set of parameters is employed to represent the codebook. Furthermore, we illustrate that the proposed scheme has an interpretation as nonlinear predictive quantization. Numerical results indicate high gain over memoryless coding and memory quantization based on linear predictive coding. The results also show that the sub-optimal approach performs close to the optimal.

关键词： Vector quantization Decoding Signal processing Encoding Block codes Zinc Sensor systems linear predictive coding Modems Speech

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Harmonic+noise coding using improved V/UV mixing and efficient spectral quantization

Harmonic+noise coding using improved V/UV mixing and efficie...

引用

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

作者： E.W.M. Yu Cheung-Fat Chan Department of Electronic Engineerin City University of Hong Kong Kowloon Hong Kong China

This paper presents a harmonic+noise speech coder which uses an efficient spectral quantization technique and a novel voiced/unvoiced (V/UV) mixing model. The harmonic magnitudes are coded at 23 bits/frame using the magnitude response of a linear predictive coding (LPC) system. The difference between the harmonic magnitudes and the sampled magnitude response is minimized by the closed-loop approach. The V/UV mixing is modeled by a smooth function which is derived from the speech spectrum envelope based on the flatness measure. The V/UV mixing model allows noise to be added in the harmonic portion of speech spectrum so that buzzyness is reduced. The V/UV mixing information is determined from the spectral parameters available in the decoder, no bits are needed for transmitting the V/UV information. A 1.4 kbps harmonic coder is developed. The speech quality of the coder is comparable to other harmonic coders operating at higher rates.

关键词： Quantization linear predictive coding Power harmonic filters Frequency Decoding Speech enhancement Speech coding Speech analysis Electronic mail Noise reduction

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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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Development of high performance DCMS with 3-bit and 4-bit coding ADPCM

Development of high performance DCMS with 3-bit and 4-bit co...

引用

IEEE International Conference on Communications (ICC)

作者： Y. Shoji O. Noguchi T. Suzuki Digital Communications Laboratories Oki Electric Industry Company Limited Tokyo Japan

A digital circuit multiplication system (DCMS), which is equipped with an echo canceller to perform activity detection by applying reciprocal action between an echo canceller and a speech detector, is described. A high-quality speech detector, developed by introducing a hangover controller and linear predictive coding (LPC) analysis, and a trial model of the DCMS using advanced adaptive differential pulse-code modulation (ADPCM) which allowed 4-bit coding for voiceband data (VBD) signals, are presented. Efficiency improved by 10% using 4-bit coding for VBD.< >

关键词： Echo cancellers Detectors Speech analysis linear predictive coding Pulse modulation Digital circuits Speech coding Adaptive signal detection Signal analysis predictive models

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System on a chip for petroleum pipeline inspection

System on a chip for petroleum pipeline inspection

引用

Symposium on Integrated Circuits and Systems Design (SBCCI)

作者： F. da Silva Dutra F. Galvez-Durand V. Alves Universidade Federal do Rio de Janeiro Brazil

The present paper describes the implementation and experimental results of the DAQP02 system on a chip (SoC) for petroleum pipeline inspection. This integrated circuit is able to read and process information about physical phenomena inside pipelines. The DAQP02 has two multiplexed analog inputs, one 8-bit A/D converter, an 8-bit data bus and a 22-bit address bus. Due to its small dimensions and low power consumption features, it is an efficient in-line inspection tool. The integrated circuit was manufactured in CMOS 0.8 /spl mu/m double poly, double metal, n-well technology.

关键词： Petroleum Pipelines Inspection System-on-a-chip Energy consumption CMOS technology Data acquisition Prototypes linear predictive coding CMOS integrated circuits

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Speaker verification using temporal decorrelation post-processing

Speaker verification using temporal decorrelation post-proce...

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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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Leak monitoring system for gas pipelines

Leak monitoring system for gas pipelines

引用

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

作者： I. Brodetsky M. Savic Electrical Computer and System Engineering Department Signal and Speech Research Laboratory Rensselaer Polytechnic Institute Troy NY USA

An approach and a solution to the continuous leak monitoring problem in underground gas pipelines are presented. This approach places permanent monitoring units along the pipeline. These units detect acoustic signals in the pipeline and discriminate leak sounds from other man-made or natural nonleak sounds that can occur. The system uses the kNN classifier as the detector with LPC (linear predictive coding) cepstrums as signal features. To increase system performance, pipeline effects on acoustic signals were taken into account during the classifier training phase. Each unit can detect 1/4-in-diameter leaks from a distance of 300 m, yielding 600 m as the maximum distance between units.< >

关键词： Monitoring Pipelines Acoustic signal detection Leak detection linear predictive coding Signal detection Neural networks Detectors Cepstrum System performance

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Snap-drift learning for phrase recognition

Snap-drift learning for phrase recognition

引用

International Joint Conference on Neural Networks (IJCNN)

作者： Sin Wee Lee D. Palmer-Brown Computational Intelligence Group School of Computing Leeds Metropolitan University Leeds UK School of Computing and Technology University of East London London UK

This paper presents a new application of the snap-drift algorithm by S. W. Lee, et al. (2004): phrase recognition using a set of phrases from the Lancaster parsed corpus (LPC) by R. Garside, et al. (1987). The learning algorithm is the classifier version of snap-drift. In this version, along with the complementary concepts of fast minimalist learning (snap) and slow drift towards the input pattern, each node of the snap-drift neural network (SDNN) swaps between snap and drift modes when declining performance is indicated on that particular node. This method enables the SDNN to learn at node level, in the sense that each node has its learning mode toggled independently of the other nodes. Learning on each node is also reinforced by enabling learning with a probability that decreases with increasing performance. The simulations demonstrate that learning is stable, and the results have consistently shown similar classification performance and advantages in terms of speed in comparison with a multilayer perceptron (MLP) and back-propagation by J. Topper, et al. (2002), D. E. Rumelhart, et al. (1986) applied to the same problem.

关键词： Subspace constraints Neural networks linear predictive coding Resonance Fuzzy set theory Computer network management Speech Computational modeling Silicon compounds Computational intelligence

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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...

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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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