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检索条件"主题词=Linear Predictive coding"

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An improved model for speech excitation using time-frequency characterisation

An improved model for speech excitation using time-frequency...

引用

International Symposium on Signal Processing and Its Applications (ISSPA)

作者： S. Ghaemmaghami M. Deriche Electronic Research Center Sharif University of Technology Tehran Iran School of Electrical and Electronic Systems Engineering Queensland University of Technology Brisbane Australia

This paper introduces a new method for approximating the excitation signal in a linear predictive coding (LPC) model over both time and frequency domains. In the frequency domain, a new two-band excitation model is proposed for decomposing the signal spectrum into two periodic and non-periodic components using the smoothed Wigner-Ville (SWV) distribution. This model provides high resolution in the time-frequency plane which gives accurate information about pitch and voicing. The periodic component is then analysed with a new "multi-pulse" like technique to approximate the signal in the time domain through repeating a representative pattern across each frame. It is shown that combining the spectral and the temporal models improves significantly over the performance of our earlier model for the excitation and gives a good approximation to the signal for most speech sounds. Evaluation results show that the proposed coding method operating at 2400 b/s outperforms current standard coders at similar rate.

关键词： Speech Time frequency analysis linear predictive coding Frequency domain analysis predictive models Signal resolution Pattern analysis Signal analysis Time domain analysis Code standards

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A new physical model for NVM data-retention time-to-failure

A new physical model for NVM data-retention time-to-failure

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Annual International Symposium on Reliability Physics

作者： B. De Salvo G. Ghibaudo G. Pananakakis B. Guillaumot P. Candelier G. Reimbold Lab. de Phys. des Composants a Semicond. ENSERG Grenoble France LPCS-ENSERG Grenoble France STMicroelectronics Grenoble France CEA/LETI Grenoble France

A new physical model for nonvolatile memory (NVM) data-retention time-to-failure is proposed. In this model, retention-time t/sub r/, namely log(t/sub r/), varies linearly with temperature T rather than with 1/T, as commonly assumed. The main consequence of the T-model is a drastic reduction of the extrapolated time-to-failure in highly temperature accelerated life-testing. Extensive experimental studies strongly support the T-model. The physical consistency of the T-model and its literature-data unifying character are demonstrated.

关键词： Temperature Threshold voltage Nonvolatile memory Acceleration Very large scale integration Dielectric losses linear predictive coding Microelectronics Performance evaluation Failure analysis

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A 6.1 to 13.3-kb/s variable rate CELP codec (VR-CELP) for AMR speech coding

A 6.1 to 13.3-kb/s variable rate CELP codec (VR-CELP) for AM...

引用

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

作者： S. Heinen M. Adratm O. Steil P. Vary W. Xu Inst. of Comm. Systems and Data Processing RWTH Aachen University of Technology Germany Rheinisch-Westfalische Technische Hochschule Aachen Aachen Nordrhein-Westfalen DE Department of Mobile Phone Development Siemens AG Munich Germany

We propose a new 6.1 to 13.3-kb/s speech codec called variable rate code-excited linear prediction (VR-CELP) for adaptive multi-rate (AMR) transmission over mobile radio channels such as GSM or UMTS. The AMR concept allows to operate with almost wireline speech quality for poor channel conditions and better quality for good channel conditions. This is achieved by dynamically splitting the gross bit rate of the transmission system between source and channel coding according to the current channel conditions. Thus the source coding scheme must be designed for seamless switching between rates without annoying artifacts. To enhance the transmission quality under very poor channel conditions, a new powerful error concealment strategy based on estimation theory is applied.

关键词： Speech coding Bit rate Speech codecs Speech analysis Vector quantization linear predictive coding Decoding Testing Iterative algorithms Encoding

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Electrical stress in N- and P-channel undoped-hydrogenated polysilicon thin film transistors (TFTs)

Electrical stress in N- and P-channel undoped-hydrogenated p...

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International Conference on Semiconductor

作者： F.V. Farmakis J. Brinis G. Kamarinos C.A. Dimitriadis V.K. Gueorguiev Tz.E. Ivanov ENSERG IL PCS Grenoble France Phelma Grenoble RhÃ´ne-Alpes FR Department of Physics University of Thessaloniki Thessaloniki Greece Institute of Solid States Physics Bulgarian Academy of Sciences Sofia Bulgaria

We applied electrical stress in P- and N-channel polysilicon thin film transistors. N-channel devices exhibit two-step degradation with hot-hole injection into the gate oxide followed by electron injection. Stress in P-channel devices provides interesting results in order to clarify the physical mechanisms occurring during device aging.

关键词： Thin film transistors Stress Hot carriers Degradation Physics Electrons Silicon Active matrix liquid crystal displays Threshold voltage linear predictive coding

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High quality audio coding using a novel hybrid WLP-subband coding algorithm

High quality audio coding using a novel hybrid WLP-subband c...

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International Symposium on Signal Processing and Its Applications (ISSPA)

作者： Yu Rongshan Ko Chi Chung Digital System and Application Laboratory Department of Electrical Engineering National University of Singapore Singapore

In this paper, a high quality audio coding scheme based on a novel hybrid algorithm combining warped linear prediction (WLP) and subband coding is proposed. In the proposed scheme, subband coding is used to quantize the WLP residual. A modified bit allocation algorithm is used for subband bit allocation so that the overall quantization noise to masking threshold ratio (NMR) in the coded signal is minimized. The proposed codec is capable of providing high quality audio output at low bit rate. Subjective tests have shown that the proposed codec is able to provide a performance comparable to that of the MPEG audio layer II while operating at a lower bit rate.

关键词： Audio coding Quantization Masking threshold Frequency Filters linear predictive coding Bit rate Nuclear magnetic resonance Codecs Humans

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Processing and classification of deformed speech using neural networks

Processing and classification of deformed speech using neura...

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Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： R. Tadeusiewicz A. Izworski W. Wszolek T. Wszolek Department of Automatics University of Mining and Metallurgy Krakow Poland

In many problems of medical diagnosis as well as therapy and rehabilitation, evaluation of the deformed speech signal quality is required. In problems of distorted speech diagnosis the regular methods of speech signal preprocessing and classification, used in speech or voice recognition, totally fail. Also the standard speech signal parametrization techniques (e.g. LPC or cepstral coefficients) cannot satisfactorily describe pathological speech because of its dissimilar phonetic and acoustic structure compared with correct speech, and also because the aim of the recognition process is totally different. In the paper a new method for processing and classification of pathologically deformed speech, based on neural networks techniques, is presented and discussed.

关键词： Speech processing Speech recognition Medical diagnosis Medical treatment Speech analysis Acoustic distortion linear predictive coding Cepstral analysis Pathology Neural networks

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Feature selection using genetics-based algorithm and its application to speaker identification

Feature selection using genetics-based algorithm and its app...

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

作者： M. Demirekler A. Haydar E.E. Eng. Department M.E.T.U. Ankara Turkey E.E. Eng. Department EMU Mersin Turkey

This paper introduces the use of genetics-based algorithm in the reduction of 24 parameter set (i.e., the base set) to a 5, 6, 7, 8 or 10 parameter set, for each speaker in text-independent speaker identification. The feature selection is done by finding the best features that discriminates a person from his/her two closest neighbors. The experimental results show that there is approximately 5% increase in the recognition rate when the reduced set of parameters are used. Also the amount of calculation necessary for speaker recognition using the reduced set of features is much less than the amount of calculation required using the complete feature set in the testing phase. Hence it is mote desirable to use the subset of the complete feature set found using the genetic algorithm suggested.

关键词： Covariance matrix Cepstral analysis Training data Speaker recognition Testing Genetic algorithms Impedance matching Speech linear predictive coding Gaussian distribution

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A basic study on talking aid and voice clarification for the hearing disorder persons

A basic study on talking aid and voice clarification for the...

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IEEE International Conference on Systems, Man and Cybernetics

作者： S. Yamaguchi Y. Sankai University of Tsukuba Tsukuba Japan Institute of Engineering Mechanics and Systems University of Tsukuba Tsukuba Japan

The voice of the handicapped who have lost their hearing is unclear. We propose a voice clarification system which consists of the voice recognition and voice clarification by monitoring the voice signal via a microphone and the shape pattern of mouth via a CCD camera, and by generating the clarified voice through the speaker. In order to recognize their voices, we adopt two methods: 1) a linear prediction coding cepstrum (LPC cepstrum), and 2) a neural network. The LPC cepstrum obtains the characteristic vector which is calculated by the frequency ingredient, and the neural network performs the pattern recognition for the typical wave shape corresponding to the each vowel. Furthermore, the image recognition system recognizes the shape pattern of mouth, based on the color information, estimation rules and membership functions. We confirm the efficiency of this total system to recognize the vowels.

关键词： Speech recognition Pattern recognition Shape Cepstrum Mouth linear predictive coding Neural networks Image recognition Auditory system Monitoring

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Text-dependent speaker identification using neural network on distinctive Thai tone marks

Text-dependent speaker identification using neural network o...

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International Joint Conference on Neural Networks (IJCNN)

作者： C. Tanprasert C. Wutiwiwatchai Sutat Sae-Tang Software and Language Engineering Laboratory National Electronics and Computer Technology CenterMinistry of Science Technology and Environment National Science and Technology Development Agency Bangkok Thailand

Presents a neural network based text-dependent speaker identification system for Thai language. linear prediction coefficients (LPC) are extracted from speech signal and formed feature vectors. These features are fed into a multilayer perceptron (MLP) neural network with backpropagation learning algorithm for training and identification processes. Five Thai tone marks are considered very closely in choosing the sentences in order to achieve the best speaker identification accuracy. Five speaking texts with each Thai tone and a mixed tone text are comparatively experimented. Average identification rate on 9 speakers achieves above 95% when using mixed tone text, and poor results occur with middle and low tone texts, which usually cause vagueness or unclear voices.

关键词： Neural networks Speech linear predictive coding Frequency Natural languages Paper technology Signal processing Backpropagation Speaker recognition Feature extraction

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A comparative analysis of the topological structures of different LPC feature-based speech models

A comparative analysis of the topological structures of diff...

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International Joint Conference on Neural Networks (IJCNN)

作者： L.R. Palomar T. Fukuda E.P. Dadios Department of Electronics & Communications Engineering De La Salle University Philippines Department of Mechano-Informatics Engineering and Department of Micro-Systems Engineering University of Nagoya Nagoya Japan Department of Manufacturing Engineering and Management De La Salle University Philippines

Describes initial experimentations done on three LPC (linear predictive coding) derived feature-based speech models: the LPC-cepstrum, the LSP (line spectral pair) and the postfilter-cepstrum (PFL). A comparative analysis of the topological structures of these models is also given. The structures are basically self-organizing feature maps which accept these models as inputs and after training, used to distinguish between isolated word utterances and speakers. A small database of 5 utterances and 4 speakers is initially used. The performance index of isolated word recognition and speaker identification for all models are calculated based on a hit-and-miss ratio and are also discussed. Experimental results reveal that the three parameters are comparable in performance. The LSP has a slight edge over the other two feature vectors in distinguishing isolated words.

关键词： Speech analysis linear predictive coding Cepstral analysis Speech recognition Speech coding predictive models Speaker recognition Audio recording Pulse modulation Nonlinear filters

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