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检索条件"主题词=Linear predictive coding"

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On an efficient decomposition of LPC excitation for producing natural sounding speech

On an efficient decomposition of LPC excitation for producin...

引用

IEEE Region 10 International Conference TENCON

作者： S.H. Leung L.F. Peng O.Y. Wong C.F. Chan A. Luk Department of Electronic Engineering City Polytechnic of Hong Kong Hong Kong China

The authors present a new method of modeling the excitation of the linear predictive coding (LPC) synthesis filter at low and medium bit rates. For the speech segments with regular patterns, the excitation is composed of two sequences of pulses. The first sequence is generated in a way similar to the classical physical model that consists of a glottal filter with thinned coefficients driven by a set of pitch pulses. Both the glottal function and pitch pulses are determined using the analysis-by-synthesis technique with the mean square criterion. The auxiliary sequence consists of a few pulses to supplement the first sequence for further reducing the mean square error. For unvoiced speech segments, multipulse excitation is simply used to drive the synthesis filter. Based on real speech analysis, the model has a gain on signal-to-noise ratio (SNR) of 2-3 dB for voiced segments over the multipulse LPC using 0.8-2.5 pulses/ms.< >

关键词： linear predictive coding Speech synthesis predictive models Nonlinear filters Bit rate Pulse generation Mean square error methods Signal synthesis Speech analysis Signal to noise ratio

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Optimal transformation of LSP parameters using neural network

Optimal transformation of LSP parameters using neural networ...

引用

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

作者： Hai Le Vu Laszlo Lois Department of Telecommunications Technical University of Budapest Budapest Hungary

ISBN: (纸本)0818679190

The intraframe correlation properties of line spectrum pair (LSP) are used to develop an efficient encoding algorithm using the Karhunen-Loeve (KL) transformation. The important nonuniform statistical characteristics of LSP frequencies are investigated. Based upon this nonuniform property the neural network based techniques for generating the transform vectors via system training are studied. Using the principal component analysis (PCA) network to decorrelate LSP coefficients, we show that these new approaches lead to as good or better distortion as compared to other methods for speech analysis-synthesis.

关键词： Neural networks Filters Frequency linear predictive coding Principal component analysis Speech analysis Karhunen-Loeve transforms Quantization Encoding Speech synthesis

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Word recognition and encoding using poles

Word recognition and encoding using poles

引用

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

作者： D. Tuffelli B. Groc E.N.S. d'Electronique et de radioélectricite Grenoble

At learning, LPC is used to get the reference poles corresponding to the words. During the recognition, the order of the filtering is variable and imposed by the dictionary. The distance between an input speech window and a dictionary speech window is computed with a method near Itakura's method but using a series of two-order inverse filtering. An improved dynamic programming is used allowing parallel computation for several words.

关键词： Encoding Dictionaries Dynamic programming Speech analysis Concurrent computing linear predictive coding predictive coding Speech coding Particle measurements Character recognition

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The effects of five voice characteristics on LPC quality

The effects of five voice characteristics on LPC quality

引用

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

作者： M. Kahn P. Garst Hewlett Packard Laboratories Palo Alto CA USA

Most published evaluations of LPC systems use only one or two speakers. Since LPC quality and intelligibility are known to depend on the speaker, this is an inadequate test of a synthesis system. We recorded eight men and nine women chosen from a speech data base of 81 speakers who were independently rated by two phoneticians for the presence or absence of the following voice characteristics: nasality, harshness, creak, whisper, and pitch extreme. The 17 talkers represented a balanced sample of strong positives or negatives of the five voice characteristics. Each speaker was recorded on one fifty word set from the Modified Rhyme Test. Monosyllabic word intelligibility tests were administered to 88 listeners (with four listeners per speaker set). Results from the intelligibility tests for different speakers show that vocal characteristics and resultant LPC quality are linked. Nasality and whisper are the most strongly correlated with a decreased LPC intelligibility.

关键词： linear predictive coding Speech synthesis Frequency Laboratories System testing Vocabulary Bit rate Speech recognition Speech analysis Spectral analysis

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Low bit rate CELP using ternary excitation codebook

Low bit rate CELP using ternary excitation codebook

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IEEE Conference and Exhibition on Global Telecommunications (GLOBECOM)

作者： S. Tavathia R. Porayath J.F. Doherty Department of Electrical Engineering and Computer Engineering Iowa State University Ames IA USA

Code excited linear predictor coders hold promise to achieve high quality speech at low bit rates. We propose a ternary excitation based CELP coder with a new structure to achieve toll quality speech at 4 kbps. Speech quality is maintained by allocating more bits for the codebook index allowing for larger codebooks which provide better speech quality as quantization levels increase. To allocate more bits for the codebook index a backward adaptive 10-th order LPC predictor is used. Regular structure of the lattice codebook and convexity of error surface have been exploited to greatly improve the efficiency of the search algorithm. The storage requirement of the codebook is eliminated by transmitting the position of three weights used in generating the ternary codebook instead of the codebook index. Speech quality obtained using the new CELP structure is studied and results compared with the LBG and the Gaussian codebooks.

关键词： Bit rate Transmitters Filters Speech coding linear predictive coding Technological innovation Quantization Lattices Pulse modulation Speech analysis

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CCD adaptive filtering for robust LPC speech processing

CCD adaptive filtering for robust LPC speech processing

引用

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

作者： G. Borsuk M. White Westinghouse Advanced Technology Laboratories Baltimore MD USA

A monolithic CCD adaptive filter chip is described which implements the Widrow-Hoff "clipped-data" LMS adaptive algorithm. The chip can be used as a pre-filter noise canceller, analysis filter, or pre-whiten... 详细信息

关键词： Charge coupled devices Adaptive filters linear predictive coding Speech processing Noise robustness Least squares approximation Adaptive algorithm Noise cancellation Speech analysis Nonlinear filters

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Objective speech quality measure for cellular phone

Objective speech quality measure for cellular phone

引用

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

作者： K.H. Lam O.C. Au C.C. Chan K.F. Hui S.F. Lau Department of Electrical and Electronic Engineering Hong Kong University of Science and Technology Kowloon Hong Kong China

Cellular phone network speech quality monitoring is a regular task performed by the cellular service providers. Objective speech quality measures are needed in such tasks to provide a reasonably accurate estimate of subjective quality of the network. We performed an experiment to collect real distorted data, conducted a survey to obtain subjective quality measure of the collected speech samples and studied the statistical correlation of 32 objective speech quality measures with the subjective measures. Four of the objective measures were found to be good. Synchronization was found to be important.

关键词： Speech Cellular phones Distortion measurement Cellular networks Time measurement Auditory system Mean square error methods linear predictive coding Frequency measurement Psychoacoustic models

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An 8 kbit/s ACELP coder with improved background noise performance

An 8 kbit/s ACELP coder with improved background noise perfo...

引用

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

作者： R. Hagen E. Ekudden Research Ericsson Radio Systems Audio and Visual Technology Sweden

This paper describes an 8 kbit/s ACELP speech coder with high performance for both speech and non-speech signals such as background noise. While the traditional waveform matching LPAS structure employed in many existing speech coders provides high quality for speech signals, it has significant performance limitations for, for example, background noise. The coder presented here employs a novel adaptive gain coding technique using energy matching in combination with a traditional waveform matching criterion providing high quality for both speech and background noise. The coder has a basic structure similar to that of the 7.4 kbit/s D-AMPS EFR coder, with a 10 th order LPC, high resolution adaptive codebook and a 4 pulse algebraic codebook. The performance for speech signals is equivalent to or better than that of state-of-the-art 8 kbit/s coders, while for background noise conditions the performance is significantly improved.

关键词： Background noise Speech enhancement Speech synthesis Speech coding Telephony Speech analysis linear predictive coding Energy resolution Signal resolution Mobile communication

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Tone modifications used for improving the discrimination of Mandarin words for the hearing-impaired

Tone modifications used for improving the discrimination of ...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Jian Lu N. Uemi Gang Li T. Ifukube Research Institute for Electronic Science Hokkaido University Sapporo Hokkaido Japan Tianjin University TJU China

In this paper, a digital processing method is described for modifying tone contrast that was defined as the difference in frequencies between peaks and valleys of pitch curves in natural utterances. Speech signals with modified tones were presented to hearing-impaired Chinese listeners who were asked to identify four alternative Mandarin words. Employing this method, it was found that modified speech with enhanced tone contrast contributed moderate gains in the percentage correct word identification when compared to unmodified speech, while reducing tone contrast generally reduced the percentage correct identification. These findings therefore offer support to the assertion that a hearing aid with tone modifications is indeed effective for hearing-impaired Chinese.

关键词： Auditory system Frequency Speech enhancement Signal processing Natural languages Speech processing linear predictive coding Vocoders Speech recognition

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Speaker-independent phonation recognition for Malay plosives using neural networks

Speaker-independent phonation recognition for Malay plosives...

引用

International Joint Conference on Neural Networks (IJCNN)

作者： Hua Nong Ting J. Yunus S.H. Salleh Faculty of Electrical Engineering Universiti Teknologi Malaysia Skudai Johor Malaysia

The paper investigates the use of neural networks in recognizing the phonation of the speech sounds. The proposed method classifies the Malay plosive sounds of adults and children based on phonation in a speaker-indep... 详细信息

关键词： Neural networks Testing Loudspeakers linear predictive coding Hidden Markov models Spatial databases Feature extraction Speech coding Speech recognition Pattern recognition

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