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检索条件"主题词=Linear Predictive Coding"

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A comparison of LSF and ISP representations for wideband LPC parameter coding using the switched split vector quantiser

A comparison of LSF and ISP representations for wideband LPC...

引用

International Symposium on Signal Processing and Its Applications (ISSPA)

作者： S. So K.K. Paliwal School of Microelectronic Engineering Griffith University Brisbane Australia

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Harmonic envelope prediction for realistic speech synthesis using kernel interpolation

Harmonic envelope prediction for realistic speech synthesis ...

引用

International Joint Conference on Neural Networks (IJCNN)

作者： P.-A. Fournier J.-J. Brault Department of Electrical Engineering École Polytechnique de Montréal Montreal QUE Canada

Harmonic and noise diphone concatenation is a proven method to obtain high-quality speech synthesis, but cannot be used when the basis corpus does not contain all the diphones needed. We propose a method to complete an individual's corpus using examples from other corpora. Parametrisation of five vowels from different speakers is done with an harmonic and noise model (HNM). We use multi-frame analysis (MFA) and smoothing kernels to estimate the harmonic power spectrum envelopes. Different kernels are compared to predict the harmonic envelopes of vowels using training data. We use euclidian distance to measure similarity between the real envelopes and the predicted ones. Synthesis of the interpolated vowels are then performed using learned optimal parameters. Our results show Gaussian kernels can achieve a 1.8 dB (34.4%) reduction of harmonic distorsion compared to the mean harmonic envelope estimator. As far as we know, there is no other literature on phoneme prediction for realistic speech synthesis.

关键词： Speech synthesis Kernel Interpolation Harmonic analysis Acoustic noise linear predictive coding Smoothing methods Power system harmonics Training data Speech coding

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A cepstral distortion measure for protein comparison and identification

A cepstral distortion measure for protein comparison and ide...

引用

International Conference on Machine Learning and Cybernetics (ICMLC)

作者： Pham Byung-Sub Shim School of Information Technology James Cook University Townsville QLD Australia Bioinformatics Applications Research Center

Protein sequence comparison is the most powerful tool for the identification of novel protein structure and function. This type of inference is commonly based on the similar sequence-similar structure-similar function paradigm, and derived by sequence similarity searching on databases of protein sequences. As entire genomes have been being determined at a rapid rate, computational methods for comparing protein sequences will be more essential for probing the complexity of molecular machines. In this paper we introduce a pattern-comparison algorithm, which is based on the mathematical concept of linear-predictive-coding based cepstral distortion measure, for comparison and identification of protein sequences. Experimental results on a real data set of functionally related and functionally non-related protein sequences have shown the effectiveness of the proposed approach on both accuracy and computational efficiency.

关键词： Cepstral analysis Distortion measurement Bioinformatics Protein sequence linear predictive coding Computational biology Protein engineering Humans Amino acids Australia

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Novel wavelet-based pitch estimation and segmentation of non-stationary speech

Novel wavelet-based pitch estimation and segmentation of non...

引用

International Conference on Information Fusion

作者： D. Charalampidis V.B. Kura EE Departinent University of New Orleans New Orleans LA USA

This paper introduces a novel method for accurate pitch estimation and speech segmentation, named multi-feature, autocorrelation (ACR) and wavelet technique (MAWT). MAWT uses feature extraction, and ACR applied on linear predictive coding (LPC) residuals, with a wavelet-based refinement step. MAWT opens the way for a unique approach to modeling: although speech is divided into segments, the success of voicing decisions is not crucial. Experiments demonstrate the superiority of MAWT in pitch period detection accuracy over existing methods, and illustrate its advantages for speech segmentation. These advantages are more pronounced for gain-varying and transitional speech, and under noisy conditions.

关键词： Feature extraction linear predictive coding Speech coding Noise robustness Cepstrum Maximum likelihood estimation Autocorrelation Speech enhancement Signal processing Frequency

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Speech synthesis from surface electromyogram signal

Speech synthesis from surface electromyogram signal

引用

IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Yuet-Ming Lam Man-Wai Mak Philip Heng-Wai Leong Department of Computer Science and Engineering Chinese University of Hong Kong Sha Tin Hong Kong China Department of Electronic and Information Engineering Hong Kong Polytechnic University Hong Kong China

This paper presents a methodology that uses surface electromyogram (SEMG) signals recorded from the cheek and chin to synthesize speech. Simultaneously recorded speech and SEMG signals are blocked into frames and transformed into features. linear predictive coding (LPC) and short-time Fourier transform coefficients are chosen as speech and SEMG features respectively. A neural network is applied to convert SEMG features into speech features on a frame-by-frame basis. The converted speech features are used to reconstruct the original speech. Feature selection, conversion methodology and experimental results are discussed. The results show that phoneme-based feature extraction and frame-based feature conversion could be applied to SEMG-based continuous speech synthesis

关键词： Speech synthesis Speech recognition Signal synthesis Network synthesis linear predictive coding Neural networks Feature extraction Fourier transforms Speech coding Working environment noise

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Java tools for teaching speech recognition

Java tools for teaching speech recognition

引用

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

作者： S. Euler Fachhochschule Gießen-Friedberg University of Applied Sciences Friedberg Germany

In this paper, we present our concept for a sequence of experiments with speech recognizers used in teaching speech recognition techniques. The experiments are performed with a combination of own tools and the hidden Markov toolkit (HTK). The first experiment demonstrates speaker dependent recognition based on the dynamic time warp algorithm. In the course of this experiment all utterances from the students are recorded and used to build up a data base. Both the recognizer and the tool used for viewing and editing the speech data are written in Java making them platform independent and easy to extend. The recorded speech data is then utilized to train and test a speaker independent recognizer.

关键词： Java Education Speech recognition Hidden Markov models Speech analysis Signal analysis Feature extraction linear predictive coding Heuristic algorithms Testing

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Machinery condition monitoring using LPC residual signal energy

Machinery condition monitoring using LPC residual signal ene...

引用

International Forum on Strategic Technology, IFOST

作者： Ui-Pil Chong Sung-Sang Lee A. Ovcharenko Sang-Jin Cho School of Computer Engineering and Information Technology University of Ulsan South Korea

Monitoring and diagnosis of the operating machines are very important for safety operation and maintenance in the industrial fields. These machines are mostly rotating machines. In this paper, we propose fault detection and diagnosis method using the LPC (linear predictive coding) and residual signal energy. We applied our method to the induction motors depending on various status of faulted condition and could obtain good results.

关键词： Machinery Condition monitoring linear predictive coding Speech analysis Equations Fault detection Power system reliability Acoustic noise Mathematical model Speech coding

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A single chip solution for text-to-speech synthesis

A single chip solution for text-to-speech synthesis

引用

European Conference on Circuit Theory and Design, ECCTD

作者： O. Aktan I.F. Baskaya G. Dundar Department of Electrical and Electronic Engineering Bogazici University Turkey Department of Electrical and Computer Engineering Georgia Institute of Technology Atlanta USA

A single chip solution for text-to-speech synthesis is presented in this paper. The proposed system is the first hardware solution for synthesizing unlimited vocabulary speech in real time for Turkish language. The integrated circuit converts incoming letters in ASCII format to speech by using clues from the textual context. The system has a language dependent database, which contains pre-recorded human speech samples coded by LPC (linear predictive coding) method. It has a very low complexity (consisting of only 3503 technology specific cells) compared to other board-level designs available in the market, resulting in low power and flexibility for FPGA implementation or incorporation into larger chips as IP.

关键词： Speech synthesis Integrated circuit synthesis Natural languages linear predictive coding Hardware Vocabulary Real time systems Databases Humans Speech coding

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Using voice conversion as a paradigm for analyzing breathy singing voices

Using voice conversion as a paradigm for analyzing breathy s...

引用

IEEE Pacific Rim Conference on Communications, Computers and Signal Processing

作者： K.I. Nordstrom G.A. Rutledge P.F. Driessen Department of Electrical and Computer Engineering University of Victoria Victoria BC Canada Research Department TC Helicon Victoria BC Canada

It is generally thought that breathiness can be added to voices by modifying the glottal source within a source-filter model. However, this does not work well when the original voice is very different from the desired breathy voice. In this experiment, a voice conversion algorithm is used to investigate the relationship between the glottal source and the vocal tract filter. The LPC residual from one voice is fed into the LPC filter of another voice. According to a source-filter theory of the voice, the synthesized voice should take on the glottal quality of the LPC source. This hypothesis is evaluated through a perceptual test with a linguistics expert. The results suggest that the vocal tract does have an influence on the perception of breathy voices. Given the narrow nature of this experiment, further testing is recommended to verify these results.

关键词： Speech analysis linear predictive coding Filters Testing Frequency Production Resonance

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Experimental evaluation of an active speech control method

Experimental evaluation of an active speech control method

引用

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

作者： K. Kondo K. Nakagawa Department of Electrical Engineering Yamagata University Yonezawa Yamagata Japan

We experimentally evaluated an active speech control scheme which reduces unnecessary speech radiated into the surrounding space. The intended application of this system, typically cellular phones, does not require speech to be radiated into the surrounding space, but only into the microphone. We previously proposed to reduce speech by generating phase-inverted predicted speech from a secondary loudspeaker. We used LPC recursively to predict samples ahead of the associated processing delay, which could go up to a few milliseconds. First, predicted samples of recorded speech were prepared off line. Then, both the original and the phase-inverted predicted samples were played out simultaneously from two loud speakers. It was found that: 1) speech cancellation of 10 dB is possible, but is highly speaker dependent; 2) the secondary loudspeaker should be oriented in the same direction as the primary source, i.e., the mouth for maximum cancellation.

关键词： Speech analysis Delay Mouth Cellular phones linear predictive coding predictive models Microphones Prediction methods Loudspeakers Position measurement

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