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检索条件"主题词=Linear predictive coding"

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Comparison of parametric and non-parametric representations of speech for recognition

Comparison of parametric and non-parametric representations ...

引用

Mediterranean Electrotechnical Conference (MELECON)

作者： O.B. Tuzun M. Demirekler K.B. Nakiboglu The Scientific and Technical Research Council of Turkey ODTÜ Ankara Electronics Research and Development Institute Ankara Turkey

In this paper, we compare several feature sets based on the parametric and non-parametric representations of speech. Parametric representations are reflection coefficients, LPC derived cepstral coefficients (CCs) and line spectral frequencies (LSFs). Non-parametric representations are based on mel-frequency cepstral coefficients (MFCCs). These different representations are evaluated by their scores of recognition, for a speaker independent, isolated word recognizer based on hidden Markov models (HMMs).< >

关键词： Speech recognition Frequency Filters Cepstral analysis Speech analysis Hidden Markov models linear predictive coding Carbon capture and storage Human voice Speech synthesis

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Vector quantization with hyper-columnar clusters

Vector quantization with hyper-columnar clusters

引用

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

作者： M. Kohata T. Takagi Faculty of Engineering University of Tohoku Sendai Japan

A new vector quantization method is proposed which can reduce search complexity and code book memory size by reducing the number of code vectors without increasing quantization distortion. This method uses hyper-columnar clusters, and an input vector is quantized to a cluster, center axis of which is nearest to the input vector. Thus, one vector and one scalar must be coded and transmitted. The proposed method was applied to four geometrically different distributions and LPC cepstrums of speech signals. As a result, the number of code vectors was decreased compared with that in an ordinary vector quantizer, in all of the above distributions. Then the reduction of memory size and search complexity were evaluated.< >

关键词： Vector quantization Narrowband Speech recognition Books linear predictive coding Cepstrum Image coding Speech coding Hidden Markov models Explosions

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Rejection for connected digit recognition based on GPD segmental discrimination

Rejection for connected digit recognition based on GPD segme...

引用

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

作者： R.A. Sukkar AT and T Bell Laboratories Inc. Naperville IL USA

Connected digit recognition is an area that has attracted significant attention because of its importance in automating a wide range of applications. In an actual application, the performance of the connected digit recognizer must be measured along two dimensions: its recognition accuracy over connected digit strings, and its rejection rate over utterances with no digits. This paper presents a rejection method for connected digit recognition. This rejection method is a post-processor to an HMM-based recognizer and consists of two stages: a digit/non-digit classification stage, and a string verification stage. The digit/non-digit classifier is discriminatively trained to determine if a given digit segment in the recognized string actually contains a digit. The string verification stage combines the results of the digit/non-digit classifier to make the final rejection decision. Experimental results on two independent databases, show that this rejection method is successful in not only rejecting speech with no connected digits, but also in rejecting putative errors, which would have resulted in misrecognition.< >

关键词： Hidden Markov models Speech recognition Cepstral analysis Databases Telephony Credit cards Robustness Monitoring Testing linear predictive coding

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Speech compression using ARMA model and wavelet transform

Speech compression using ARMA model and wavelet transform

引用

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

作者： Sun-Won Park EECS Department Texas A and M University Kingsville Kingsville TX USA

ISBN: (纸本)0780317750

In LPC (linear predictive coding) vocoding, because of the insufficiency of the all-pole model, reconstructed speech is often crude unless much information about the residual error is also transmitted. To improve the quality, ARMA parameters of each bandpass filter output of a filter bank are extracted and transmitted in this paper. A general idea of the wavelet transform is utilized for design of the filter bank. Both the ARMA model and the wavelet transform are well known. But a combination of these technique has not been exploited in speech coding. This paper may lead to the new way for the very low-rate speech coding.< >

关键词： Wavelet transforms Least squares approximation Speech coding Poles and zeros Parameter estimation Iterative algorithms linear predictive coding Filter bank Pulse modulation Computational complexity

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Design and implementation of a programmable bit-rate multipulse excited LPC vocoder for digital cellular radio applications

Design and implementation of a programmable bit-rate multipu...

引用

IEEE International Conference on Personal Wireless Communication

作者： D.V. Poornaiah P.V.A. Mohan S.M. Chadchan DSP Group Indian Telephone Ind. Bangalore India DSP Group Transmission R D Indian Telephone Industries Limited Bangalore India Department of EGE Karnataka Regional Engineering College India

We present the algorithm for design of an analysis-by-synthesis based multi-pulse excited/long-term prediction type LPC speech coder and discuss its real-time implementation using the state-of-art ADSP 2101 digital signal processor. The important features of this speech coder are attributed to the use of (i) an efficient search algorithm for computing the pulse locations and amplitudes involved, (ii) sigma-delta based 16-bit codec-filter VLSI chip: AD 28msp 0.2 acting as front and back ends (iii) perceptual weighting filter and (iv) an efficient output post-processing filter. A detailed objective and subjective evaluation of the speech coder is carried out.

关键词： Speech analysis Signal processing algorithms Algorithm design and analysis Filters Signal design Signal analysis linear predictive coding Speech processing Digital signal processors Speech codecs

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Speaker recognition in tactical communications

Speaker recognition in tactical communications

引用

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

作者： R. Ricart J. Cupples L. Fenstermacher Bcuz-Allen and Hamilton Inc. McLean VA USA Rome Laboratory NY USA

Tactical communications are inherently short and exhibit a great deal of channel variability. A novel speaker recognition technique is described in which on-line training is utilized to circumvent the need for excessive speaker or channel modeling. The technique incorporates both feature set fusion and classifier fusion. Separate classifiers are trained for each feature set: LPC cepstra with and without RASTA filtering concomitant with delta and acceleration cepstra. The results of the individual are then adjudicated to the correct speaker. The speaker recognition algorithm was baselined with the KING database, used extensively in speaker recognition. A subsequent evaluation, conducted on the Rome Laboratory GREENFLAG tactical communications database, resulted in 93% correct identification of 41 speakers.< >

关键词： Speaker recognition Acceleration Hidden Markov models Testing linear predictive coding Spatial databases Laboratories Speech Filtering USA Councils

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Speech compression for audio description on television

Speech compression for audio description on television

引用

International Conference on Storage and Recording Systems, 1994.

作者： M.R. Suddle A.M. Kondoz B.G. Evans Centre for Satellite Engineering Research University of Surrey Guildford Surrey UK

The authors have provided an overview of the 7.6 kb/s narrowband and 13.8 kb/s wideband coders they have developed for the AUDETEL application. Both of these coders are based on CELP principles and employ the authors recently developed pulsed residual excitation (PRE) codebook. This codebook is not only computationally less demanding (as compared to a centreclipped stochastic codebook) but also offers better performance. Informal listening tests have indicated that the output speech quality of the 7.6 kb/s narrowband coder is very close to 64 kb/s PCM, and that of the 13.8 kb/s wideband version is significantly higher than that of the narrowband version.< >

关键词： Data compression linear predictive coding Speech coding TV

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A multimodal, keyword-based spoken dialogue system-MultiksDial

A multimodal, keyword-based spoken dialogue system-MultiksDi...

引用

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

作者： H. Matsu'ura Y. Masai J. Iwasaki S. Tanaka H. Kamio T. Nitta Multimedia Engineering Laboratory Toshiba Corporation Kawasaki Japan

In this paper, a multimodal, keyword-based, spoken dialog system ("MultiksDial") is described. The system provides multiple input channels of spontaneous speech and designation by touch, as well as multiple output channels of graphics and voice responses by text-to-speech. The system also provides three types of sensors to detect the user's actions and to plan interactive strategies. A word spotter handles real-time word spotting on medium-sized vocabulary words. The multimodal interaction mechanism is evaluated on a directory guidance system. The experimental results showed comparative merits of usability against an ordinary touch-screen system. Multimodal input is especially effective for novice users. Cooperative guidance using sensors, graphics, and speech also helped novice users. A multimodal UI development tool has been developed for rapid prototyping of MultiksDial.< >

关键词： Speech recognition Noise cancellation linear predictive coding Graphics Working environment noise Sensor systems Prototypes User interfaces Silicon compounds Speech synthesis

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A multiscale stochastic modeling approach to the monitoring of mechanical systems

A multiscale stochastic modeling approach to the monitoring ...

引用

IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： K.C. Chou L.P. Heck Applied Control and Signal Processing Group SRI International Inc. Menlo Park CA USA

Presents results on using a statistical model motivated by the wavelet transform to represent non-stationary signals typically encountered in machinery monitoring applications. The authors propose the use of a frame-based system in which the data in each frame is modeled as a multiscale stochastic process. The parameters of a multiscale model are used as features for each frame, where each frame of features is modeled as a sample of a multivariate, multimodal distribution. Classification of machine states based on monitoring signals is performed by comparing likelihood scores for each machine state. The authors present an example of applying the system to data consisting of a superposition of damped sinusoids, as a way of illustrating system performance for the case of transient monitoring signals. They compare their system to one which is trained using a DFT-based (non-time-frequency-based) representation (in particular, LPC coefficients) and show that their system exhibits both superior performance as well as greater robustness to noise in the signals. They also compare results using multiscale parameters versus LPC coefficients for the case of synthesized autoregressive signals and for the case of actual, measured signals from a weld depth monitoring system.< >

关键词： Stochastic systems Mechanical systems Condition monitoring linear predictive coding Machinery Time frequency analysis Binary trees Process control Signal processing Wavelet transforms

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Vector quantization-lattice vector quantization of speech LPC coefficients

Vector quantization-lattice vector quantization of speech LP...

引用

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

作者： Jianping Pan T.R. Fischer School Electrical Engineering and Computer Science Washington State University Pullman WA USA

Two-stage vector quantization-lattice vector quantization (VQ-LVQ) is used to encode the speech line spectrum pair (LSP) parameters. VQ-LVQ has the advantages of lower implementational complexity and less required memory than split vector quantization (SVQ) and multi-stage vector quantization (MSVQ) with unstructured codebooks. Based on the authors' speech data base and the same spectral measure, VQ-LVQ can save about 3 bits/frame compared to SVQ, and has advantages of about 2 to 3 bits/frame compared to unstructured codebook MSVQ, depending on the number of stages and the survivor path search complexity. The paper also provides a discussion on some factors influencing the evaluation of the LSP encoding performance.< >

关键词： Vector quantization linear predictive coding Encoding Distortion measurement Speech analysis Speech coding Redundancy Lattices Computer science Reflection

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