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检索条件"主题词=Linear predictive coding"

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Automatic classification of singing voice quality

Automatic classification of singing voice quality

引用

International Conference on Intelligent Systems Design and Applications (ISDA)

作者： B. Kostek P. Zwan Multimedia Systems Department Gdańsk University of Technology Poland

In the paper problems related to the classification of singing voice quality are presented. For this purpose a database consisting of singers' sample recordings is constructed and parameters are extracted from recorded voice of trained and untrained singers. The parameterization process is based on both voice source and formant analysis of a singing voice. These parameters are explained as to their physical interpretation and analyzed statistically in order to diminish their number. The statistical analysis is based on the Fisher statistic. In such a way a feature vector of a singing voice is formed. Decision systems based on neutral networks and rough sets are utilized in the context of the voice type and voice quality classification. Results obtained in the automatic classification performed by both decision systems are compared. A possibility to classify automatically type/quality of voice is judged. The methodology proposed provides means for discerning trained and untrained singers.

关键词： Frequency Statistical analysis Speech analysis Rough sets Biomechanics Intelligent systems Signal analysis linear predictive coding Paper technology Multimedia systems

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Spectral stability based event localizing temporal decomposition

Spectral stability based event localizing temporal decomposi...

引用

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

作者： A.C.R. Nandasena M. Akagi Graduate School of Information Science Japan Advanced Institute of Science and Technology Japan

A new approach to temporal decomposition (TD) of speech, called "spectral stability based event localizing temporal decomposition", abbreviated S/sup 2/ BEL-TD, is presented. The original method of TD proposed by Atal (1983) is known to have the drawbacks of high computational cost, and the instability of the number and locations of events. In S/sup 2/ BEL-TD, the event localization is performed based on a maximum spectral stability criterion. This overcomes the instability problem of events of the Atal's method. Also, S/sup 2/ BEL-TD avoids the use of the computationally costly singular value decomposition routine used in the Atal's method, thus resulting in a computationally simpler algorithm of TD. Simulation results show that an average spectral distortion of about 1.5 dB can be achieved with LSF as the spectral parameter. Also, we have shown that the temporal pattern of the speech excitation parameters can also be well described using the S/sup 2/ BEL-TD technique.

关键词： Speech Singular value decomposition Information science Costs Computational modeling linear predictive coding Solids Stability criteria

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Pitch Synchronous LPC Analysis for Reduction of coding Distortion caused by Misalignment between Synthesis Filter and its Residual

Pitch Synchronous LPC Analysis for Reduction of Coding Disto...

引用

International Conference on Information, Communications and Signal Processing

作者： H. Farsi Department of Electronic and Electrical Engineering Faculty of Engineering University of Birjand Birjand Iran

Several pre-processing algorithms modify the residual speech signal to facilitate efficient estimation of speech model parameters. This, however, can result in misalignment between the modified residual signal and the time-variant linear prediction (LP) filter used during the synthesis stage. The resulting misalignment may cause audible artifacts particularly at onsets when the frequency response of successive LP filters changes rapidly. We propose a new solution to control the LP filter gain at subframes. This technique is performed before and after time modification of speech and therefore is called preanalysis and post-processing. A pitch smoothing technique is used to illustrate the effect of the proposed technique

关键词： linear predictive coding Speech synthesis Signal synthesis Nonlinear filters Pulse amplifiers Speech coding Pulse measurements Delay Speech analysis Distortion

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Efficient algorithm to compute LSP parameters from 10th-order LPC coefficients

Efficient algorithm to compute LSP parameters from 10th-orde...

引用

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

作者： S. Grassi A. Dufaux M. Ansorge F. Pellandini Institute of Microtechnology University of Neuchâtel Neuchatel Switzerland

Line spectrum pair (LSP) representation of linear predictive coding (LPC) parameters is widely used in speech coding applications. An efficient method for LPC to LSP conversion is Kabal's method. In this method the LSPs are the roots of two polynomials P'/sub p/(x) and Q'/sub p/(x), and are found by a zero crossing search followed by successive bisections and interpolation. The precision of the obtained LSPs is higher than required by most applications, but the number of bisections cannot be decreased without compromising the zero crossing search. In this paper, it is shown that, in the case of 10th-order LPC, five intervals containing each only one zero crossing of P'/sub 10/(x) and one zero crossing of Q'/sub 10/(x) can be calculated, avoiding the zero crossing search. This allows a trade-off between LSP precision and computational complexity resulting in considerable computational saving.

关键词： linear predictive coding Polynomials Interpolation Speech coding Computational complexity Filters Stability Narrowband Chebyshev approximation Frequency

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Speech scrambling prior to LPC coding

Speech scrambling prior to LPC coding

引用

IEE Colloquium on Security and Cryptography Applications to Radio Systems

作者： C.S. Xydeas D.J. Hiotakakos C.A. Boyd Speech of Processing Research Laboratory Electrical Engineering Department University of Manchester Institute of Science and Technology UK

Source coding and encryption are linked theoretically by the aim of removing redundancy. However, so far, no attempt has been made to combine source coding and encryption together, except in lossless compression models. When applied to speech coding, this appears to be a novel idea for research which provides the potential for developing new designs and implementations. The authors have considered the effect of combined encoding and encryption on analysis-by-synthesis (AbS) LPC based techniques, a class of time domain speech compression algorithms which are widely used nowadays in commercial as well as military communication applications. A novel pre-processing speech scrambling algorithm (PSSA) is proposed, which given speech, produces a scrambled signal with speech like characteristics. The resulting signal can be then compressed by a low bit rate speech codec.< >

关键词： Cryptography linear predictive coding Speech coding Speech synthesis

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A Formant Frequency Estimation Scheme for Speech Signals in the Presence of Noise

A Formant Frequency Estimation Scheme for Speech Signals in ...

引用

URSI International Symposium on Signals, Systems, and Electronics (ISSSE)

作者： S.A. Fattah W.-P. Zhu M.O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, a new scheme for the estimation of formant frequencies of noise-corrupted speech signals is presented. A once-repeated autocorrelation function (ORACF) of the observed noisy speech signal is proposed to employ in a linear predictive based formant estimation method. It has been shown that the ORACF is capable of reducing the effect of additive noise significantly and if, instead of conventional ACF, ORACF is used in a modified form of least-squares Yule-Walker equations, a better performance in the formant estimation is achieved. Moreover, a frequency-domain algorithm is incorporated in the proposed scheme to avoid the possible estimation error in extracting a formant with low energy. The proposed algorithm has been tested on synthetic and natural vowels as well as some naturally spoken sentences in the presence of additive noise. The experimental results demonstrate a better performance obtained by the proposed scheme in comparison to some of the existing methods at low levels of signal-to-noise ratio (SNR).

关键词： Frequency estimation Speech enhancement linear predictive coding Speech analysis Working environment noise Autocorrelation Additive noise Speech synthesis Signal processing Equations

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Speech synthesis using two-sided linear prediction parameters

Speech synthesis using two-sided linear prediction parameter...

引用

International Symposium on Speech, Image Processing and Neural Networks

作者： S.H. Leung H.C. Ng K.F. Wong Department of Electronic Engineering City Polytechnic of Hong Kong Hong Kong China

A two-sided linear prediction (TSLP) model is shown to have high prediction gain over the conventional linear prediction (LPC) model [David and Ramamurthi, 1991], while it requires fewer coefficients in modeling. Unfortunately, speech synthesis cannot use the TSLP model directly because it needs future samples which are not available in the process. Autoregressive spectral matching (ARSM) is proposed to render the TSLP model suitable for speech synthesis. Vector sum excitation method is used to generate the excitation to the new model and its performance is comparable to the standard VSELP.< >

关键词： Speech synthesis predictive models Power system modeling linear predictive coding Autocorrelation Equations Speech processing Image processing Neural networks Cities and towns

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A speaker independent isolated turkish syllable recognition system

A speaker independent isolated turkish syllable recognition ...

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IEEE Signal Processing and Communications Applications (SIU)

作者： N. Ikizler I.H. Cavdar G. Ekim Karadeniz Teknik Üniversitesi Elektronik Mühendisliği Bölümü Trabzon Turkey

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Very low data rate speech compression with LPC vector and matrix quantization

Very low data rate speech compression with LPC vector and ma...

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

作者： D. Wong B. Juang D. Cheng Speech Technology and Research Laboratory Santa Barbara CA USA Bell Laboratories Holmdel NJ USA University of California Santa Barbara CA USA

Frame predictive vector quantization is developed to compress the bit rate for coding the LPC filter coefficients to under 250 bits/sec. An innovative LPC compression technique, matrix quantization, is also developed to compress the LPC filter coefficients to a rate under 150 bits/sec. Subjective evaluation with the diagnostic rhyme test (DRT) finds the proposed techniques to be feasible for intelligible speech transmission at bit rates between 400 bits/sec and 200 bits/sec.

关键词： linear predictive coding Bit rate Vector quantization Testing Filters Speech synthesis Speech analysis Vocoders Encoding Entropy

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Rule synthesis of speech from dyadic units

Rule synthesis of speech from dyadic units

引用

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

作者： J. Olive Bell Laboratories Murray Hill NJ USA

Peterson, Wang, and Sivertsen[1] suggested the use of the units called "dyads" as the basic unit for speech synthesis. This paper describes an approach to speech synthesis by rule which uses a unit that is similar but smaller than the dyad as defined by Peterson et al. This new unit specifies only the transition between the two phones of the dyad, while the "steady state" portions are obtained by connecting with straight lines the end points of adjacent transitions. Further simplifications of the dyadic concept include a reduced collection of dyadic transitions, and the storage of only the end points of the dyadic transitions: the transitions themselves are then obtained by interpolation between these end points. This paper describes a complete rule synthesis scheme which uses these simplified dyads in combination with a word pronouncing dictionary and suitable prosodic rules.

关键词： Speech synthesis Dictionaries Steady-state Smoothing methods Joining processes Speech recognition linear predictive coding

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