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Linear predictive coding

About: Linear predictive coding is a research topic. Over the lifetime, 6565 publications have been published within this topic receiving 142991 citations. The topic is also known as: Linear predictive coding, LPC.


Papers
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Patent
Shihua Wang1
19 Oct 1999
TL;DR: In this paper, a speech encoding method using analysis-by-synthesis includes sampling an input speech and dividing the resulting speech samples into frames and subframes, the frames are analyzed to determine coefficients for the synthesis filter.
Abstract: A speech encoding method using analysis-by-synthesis includes sampling an input speech and dividing the resulting speech samples into frames and subframes. The frames are analyzed to determine coefficients for the synthesis filter. The subframes are categorized into unvoiced, voiced and onset categories. Based on the category, a different coding scheme is used. The coded speech is fed into the synthesis filter, the output of which is compared to the input speech samples to produce an error signal. The coding is then adjusted per the error signal.

37 citations

Journal ArticleDOI
TL;DR: This paper presents an interpretation of the log likelihood ratio measure within the theoretical framework of a waveform coder distortion model, and discusses the implications of this interpretation and how it can be applied to the formulation of better objective measures of wave form coder performance.
Abstract: The log likelihood measure has been widely used in speech research for comparing speech signals. Recently, it has been proposed as a measure for assessing the quality of coded speech. In this paper we present an interpretation of the log likelihood ratio measure within the theoretical framework of a waveform coder distortion model. We then discuss the implications of this interpretation and show how it can be applied to the formulation of better objective measures of waveform coder performance.

37 citations

Journal ArticleDOI
TL;DR: A new method for VFR using the norm of the derivative parameters in deciding to retain or to discard a frame is introduced, and informal inspection of speech spectrograms shows that this new method puts more emphasis on the transient regions of the speech signal.
Abstract: Variable frame rate (VFR) analysis is a technique used in speech processing and recognition for discarding frames that are too much alike. The article introduces a new method for VFR. Instead of calculating the distance between frames, the norm of the derivative parameters is used in deciding to retain or to discard a frame, informal inspection of speech spectrograms shows that this new method puts more emphasis on the transient regions of the speech signal. Experimental results with a hidden Markov model (HMM) based system show that the new method outperforms the classical method. >

37 citations

Proceedings ArticleDOI
21 Apr 1997
TL;DR: A novel low-delay wideband speech coder that employs a multi-band bank of off-line filtered excitation codebooks, fullband linear prediction synthesis, and minimization of the error between the original and the synthesized speech signal over the full frequency range is described.
Abstract: A novel low-delay wideband speech coder, called multiband CELP (MB-CELP), overcomes the major obstacles usually associated with two traditional CELP approaches to wideband speech coding-namely fullband CELP and split-band CELP. The new MB-CELP coder employs a multi-band bank of off-line filtered excitation codebooks, fullband linear prediction synthesis, and minimization of the error between the original and the synthesized speech signal over the full frequency range. A 16 kbps version of the MB-CELP coder with two equal bands, is described. Subjective comparison test results show that this coder performs better than the G.722 coder at a bit-rate of 48 kbps.

37 citations

Journal ArticleDOI
TL;DR: If the speech correlation is properly estimated and the previously derived subband filters discussed in this work show significantly less speech distortion compared to conventional noise reduction algorithms, the quality and intelligibility of the processed signals are predicted by objective measures.
Abstract: Recently, it has been proposed to use the minimum-variance distortionless-response (MVDR) approach in single-channel speech enhancement in the short-time frequency domain. By applying optimal FIR filters to each subband signal, these filters reduce additive noise components with less speech distortion compared to conventional approaches. An important ingredient to these filters is the temporal correlation of the speech signals. We derive algorithms to provide a blind estimation of this quantity based on a maximum-likelihood and maximum a-posteriori estimation. To derive proper models for the inter-frame correlation of the speech and noise signals, we investigate their statistics on a large dataset. If the speech correlation is properly estimated, the previously derived subband filters discussed in this work show significantly less speech distortion compared to conventional noise reduction algorithms. Therefore, the focus of the experimental parts of this work lies on the quality and intelligibility of the processed signals. To evaluate the performance of the subband filters in combination with the clean speech inter-frame correlation estimators, we predict the speech quality and intelligibility by objective measures.

37 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
20239
202225
202126
202042
201925
201837