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检索条件"主题词=Linear predictive coding"

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GSM to G.729 speech transcoder

GSM to G.729 speech transcoder

引用

IEEE International Conference on Electronics, Circuits and Systems (ICECS)

作者： Shu-Min Tsai Jar-Ferr Yang Department of Electrical Engineering National Cheng Kung University Taiwan ROC Department of Electrical Engineering National Chiao Tung University Taiwan

With increasing demand of wireless and Internet accesses, the interoperability crossing these two networks becomes increasingly important for modern communications. A transcoding system that translates the coding parameters directly between the GSM and G.729 speech coders is discussed in the present paper. The GSM coder is used for mobile communications while the G.729 is the most favorable speech coder in Internet communications. For Internet/wireless gateway servers, it is evident that the proposed transcoding system requires much less computation than the conventional decode-then-encode (DTE) approach.

关键词： GSM Transcoding Speech coding Mobile communication Decoding Internet telephony Filters linear predictive coding IP networks Switches

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Comparative analysis of speech parameters for the design of speaker verification systems

Comparative analysis of speech parameters for the design of ...

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

作者： A.F. Souza M.N. Souza Biomedical Engineering Program-COPPE Federal University of Rio de Janeiro Rio de Janeiro Brazil Electronic Department E.E. Federal University of Rio de Janeiro Rio de Janeiro Brazil

Speaker verification systems are basically composed of three stages: feature extraction, feature processing and comparison of the modified features from speaker voice and from the voice that should be verified. Many features have been used in the first stage, although the current literature has not already shown the best of them. Based on the biometrics hypothesis, which states that each individual has a physical characteristic that distinguishes itself from the others, this paper realized a comparison between 12 classical widely used parameters, in order to investigate the biometrics hypothesis. The obtained results point out those parameters directly correlated to speaker's anatomy which are among the best ones that can be used in the development of speaker verification systems.

关键词： Speech analysis Biometrics Covariance matrix Forensics Speaker recognition Feature extraction Robustness predictive models linear predictive coding Biomedical engineering

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Design and evaluation of a voice conversion algorithm based on spectral envelope mapping and residual prediction

Design and evaluation of a voice conversion algorithm based ...

引用

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

作者： A. Kain M.W. Macon Center for Spoken Language Understanding CSLU Oregon Graduate Institute Beaverton OR USA

The purpose of a voice conversion (VC) system is to change the perceived speaker identity of a speech signal. We propose an algorithm based on converting the LPC spectrum and predicting the residual as a function of the target envelope parameters. We conduct listening tests based on speaker discrimination of same/difference pairs to measure the accuracy by which the converted voices match the desired target voices. To establish the level of human performance as a baseline, we first measure the ability of listeners to discriminate between original speech utterances under three conditions: normal, fundamental frequency and duration normalized, and LPC coded. Additionally, the spectral parameter conversion function is tested in isolation by listening to source, target, and converted speakers as LPC coded speech. The results show that the speaker identity of speech whose LPC spectrum has been converted can be recognized as the target speaker with the same level of performance as discriminating between LPC coded speech. However, the level of discrimination of converted utterances produced by the full VC system is significantly below that of speaker discrimination of natural speech.

关键词： Algorithm design and analysis Testing Virtual colonoscopy linear predictive coding Databases Natural languages Humans Speech recognition Target recognition Loudspeakers

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A feature independent of bit rate for twinvq audio retrieval

A feature independent of bit rate for twinvq audio retrieval

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： K. Onishi M. Kobayakawa M. Hoshi T. Ohmori Graduate School of Information Systems University of Electro-Communications Chofu Tokyo Japan

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A real-time 64-monosyllable recognition LSI with learning mechanism

A real-time 64-monosyllable recognition LSI with learning me...

引用

Asia and South Pacific Design Automation Conference

作者： K. Nakamura Qiang Zhu S. Maruoka T. Horiyama S. Kimura K. Watanabe Graduate School of Information Science Nara Institute of Science and Technology Ikoma Nara Japan Fujitsu Laboratories Limited Japan Hitachi and Limited Japan

ISBN: (纸本)0780366336

In the paper, a real-time 64-mono-syllable recognition LSI is presented. The LSI accepts 11.6 ms speech frame and outputs a 6-bit symbol-code for each frame by the end of the next frame in a pipelining manner. The recognition method is based on the Hidden Markov Model (HMM) and is speaker-independent. An on-chip learning mechanism has also been designed, but the circuit is off-chip for the present implementation because of the restriction of LSI area. The LSI is fabricated by VDEC Rohm with a 0.6 /spl mu/m CMOS process on a 4.5 mm/spl times/4.5 mm chip.

关键词： Large scale integration Learning systems Hidden Markov models Speech recognition linear predictive coding Circuits Speech coding Hardware Speech analysis Adders

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Soft computing design for speech-based diagnosis of the human throat

Soft computing design for speech-based diagnosis of the huma...

引用

Conference of the North American Fuzzy Information Processing Society - NAFIPS

作者： M.P. Beddoes Department of Electrical and Computer Engineering University of British Columbia Canada

The speech wave passes through the glottis and the vocal tract and finally arrives at the lips. The shapes of these organs modify the sounds we hear. A "tube model" can represent the vocal tract, and the various cross-sections progressing from the glottis to the lips characterize differing sounds. Such cross-sections may be linked to patient state. This may be NORMAL or ABNORMAL, damaged by cancer or unfortunate surgery or just old age. The author is interested in using vowel sounds as a possible diagnostic tool. Cross-sections of the vocal tract model can be calculated from the constants of a linear prediction coder, LPC, trained with a suitable vowel. The paper shows that an artificial neural network version of the LPC gives lower error than the original which should make it more useful for calculating cross-sectional areas. Two stages of fuzzy variables are used to diagnose a patient into two levels, NORMAL or ABNORMAL. Further analysis reinforces the diagnosis by using adverbs LIKELY or UNLIKELY to end up with one of four categories.

关键词： Humans linear predictive coding Lips Artificial neural networks Speech Frequency selective surfaces Shape Cancer Oncological surgery predictive models

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Evaluation of mel-LPC cepstrum in a large vocabulary continuous speech recognition

Evaluation of mel-LPC cepstrum in a large vocabulary continu...

引用

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

作者： H. Matsumoto M. Moroto Faculty of Engineering Shinshu University Nagano Nagano Japan

This paper presents a simple and efficient time domain technique to estimate an all-pole model on the mel-frequency scale (mel-LPC), and compares the recognition performance of the mel-LPC cepstrum with those of both the standard LPC mel-cepstrum and the MFCC (mel-frequency cepstral coefficient) through the Japanese dictation system (Julius) with 20,000 word vocabulary. First, the optimal value of the frequency warping factor is examined in terms of monosyllable accuracy. When using the optimal warping factors, the mel-LPC cepstrum attains word accuracies of 93.0% for male speakers and 93.1% for female speakers, which are 2.1% and 1.7% higher than those of the LPC mel-cepstrum, respectively. Furthermore, this performance is slightly superior to that of MFCC.

关键词： Cepstrum Vocabulary Speech recognition linear predictive coding Cepstral analysis Psychoacoustic models Mel frequency cepstral coefficient Frequency conversion Automatic speech recognition Spectral analysis

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Neural-network-based HMM adaptation for noisy speech

Neural-network-based HMM adaptation for noisy speech

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

作者： S. Furui D. Itoh Department of Computer Science Tokyo Institute of Technology Meguro Tokyo Japan

This paper proposes a new method, using neural networks, of adapting phone HMMs to noisy speech. The neural networks are designed to map clean speech HMMs to noise-adapted HMMs, using noise HMMs and signal-to-noise ratios (SNRs) as inputs, and are trained to minimize the mean square error between the output HMMs and the target noise-adapted HMMs. In evaluation, the proposed method was used to recognize noisy broadcast-news speech in speaker-dependent and -independent modes. The trained networks were confirmed to be effective in recognizing new speakers under new noise and various SNR conditions.

关键词： Hidden Markov models Speech enhancement Signal to noise ratio Neural networks Additive noise Gaussian noise linear predictive coding Speech recognition Computer science Signal design

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Pitch-excited ARMA lattice model for speech synthesis

Pitch-excited ARMA lattice model for speech synthesis

引用

IEEE Pacific Rim Conference on Communications, Computers and Signal Processing

作者： H.K. Kwan M. Wang Department of Electrical and Computer Engineering University of Windsor Windsor ONT Canada

ISBN: (纸本)0780370805

We present the ARMA lattice model for speech synthesis. By adopting a pole-zero approach, this model overcomes the limitation of the absence of zeros in the LPC model, which is based entirely on an all-pole AR model for the representation of the vocal tract. Therefore, a more natural synthesized speech can be achieved using this ARMA approach, especially in speech signals that contain nasal, fricative and plosive sounds. The structure of the ARMA lattice model is numerically stable, which is desirable for robust speech synthesis applications. Simulation results indicate that the quality of the synthesized speech of spoken sentences is encouraging, and a better result can be achieved by optimizing the parameters to be used.

关键词： Lattices Speech synthesis Poles and zeros Filters linear predictive coding Signal synthesis Speech analysis Numerical models Robustness Resonance

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Maximum-take-precedence ACELP: a low complexity search method

Maximum-take-precedence ACELP: a low complexity search metho...

引用

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

作者： Fu-Kun Chen Jar-Ferr Yang Department of Electrical Engineering National Cheng Kung University Tainan County Taiwan

The ACELP method makes use of multipulse structure to represent the excitation pulses of residual signal. With the purpose of computational complexity reduction, this paper provides the maximum-take-precedence ACELP (MTP-ACELP) search method under the acceptable degradation in performance. Because the maximum of target signal is preferentially compensated, the degradation of performance would be diminished. By predicting the locations of pulses, the computational complexity would be reduced. We not only reduce the possible pulse combinations in the search procedure but also avoid the computation of useless correlation functions before the search procedure. Furthermore, the proposed method is compatible to any ACELP type vocoder, e.g. the G.723.1, G.729, GSM- EFR standards.

关键词： Search methods Computational complexity linear predictive coding Speech coding Stochastic processes Speech analysis Covariance matrix Degradation Speech codecs Signal synthesis

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