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Code-excited linear prediction

About: Code-excited linear prediction is a research topic. Over the lifetime, 2025 publications have been published within this topic receiving 28633 citations. The topic is also known as: CELP.


Papers
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Journal ArticleDOI
TL;DR: A new technique for the class of code-excited linear prediction speech codecs designed to reduce error propagation after lost frames is presented, which consists in replacing the interframe long-term prediction with a glottal-shape codebook in the subframe containing the firstglottal impulse in a given frame.
Abstract: This paper presents a new technique for the class of code-excited linear prediction speech codecs designed to reduce error propagation after lost frames. Its principle consists in replacing the interframe long-term prediction with a glottal-shape codebook in the subframe containing the first glottal impulse in a given frame. This technique, independent of previous frames, is of particular interest in voiced speech frames following transitions as these frames are the most sensitive to frame erasures. It is a basis of a structured coding scheme called transition coding (TC). The TC greatly improves codec performance in noisy channels while maintaining clean channel performance. It is a part of the new embedded speech and audio codec recently standardized as Recommendation G.718 by ITU-T.

9 citations

01 Jan 1988
TL;DR: Because most of the computation is performed on the baseband, which is decimated, and during code-book search only the pitch filter response is considered, the complexity of CELP-BB is very much less than that of C ELP and its speech quality is not affected by LPC quantization error to such an extent as in CELE.
Abstract: y CELP [l] has proved that it is possible to use vector ions as large as 40 or more samples long and still maingh quality. This is achieved by error minimization n the original and the synthesized vectors, rather than inimization between the residual vectors and code-book in earlier VQ designs. The CELP design however, is ex and its quality is affected very much by the n error of the LPC parameters [2] making it difficult new vector quantized and easily implementer (CELP-BB) which was originally proposed below 4.8 Kb/s in [3]. Because most of the rformed on the base-band which is decimated, and that during code-book search only the pitch filter response -BB is very much less affected by LPC quantLP. CELP-BB can proo the original) down to ty starts to deteriorate natural and intelligible

9 citations

Proceedings Article
01 Jan 2006
TL;DR: This paper presents a method to handle the pitch phase error that is generally introduced by the concealment procedure of CELP speech decoders in such a way that the natural pitch periodicity of the speech signal is not broken.
Abstract: The concealment procedure used by CELP speech decoders to regenerate lost frames introduces an error that propagates into the following frames. Within the context of voice transmission over packet networks, some packets arrive too late to be decoded and must also be concealed. Once they arrive however, those packets can be used to update the internal state of the decoder, which stops error propagation. Yet, care must be taken to ensure a smooth transition between the concealed frame and the following “updated” frame computed with properly updated internal states. During voiced or quasi-periodic segments, the pitch phase error that is generally introduced by the concealment procedure makes it difficult and detrimental to quality to use the traditional fade-in, fade-out approach. This paper presents a method to handle that pitch phase error. Specifically, the transition is done in such a way that the natural pitch periodicity of the speech signal is not broken. Index Terms: speech coding, robustness, late packets

9 citations

Dissertation
01 Jan 1995
TL;DR: Results are presented which show that the variable-rate CELP speech coder for implementation on the TMS320C51 Digital Signal Processor obtains near equivalent quality compared with an 8 kbit/s fixed-rate system and significantly better quality than a fixedrate system with the same average rate.
Abstract: In a typical voice codec application, we wish to maximize system capacity while at the same time maintain an acceptable level of speech quality. Conventional speech coding algorithms operate at fixed rates regardless of the input speech. In applications where the system capacity is determined by the average rate, better performance can be achieved by using a variable-rate codec. Examples of such applications are CDMA based digital cellular and digital voice storage. . In order to achieve a high quality, low average bit-rate Code Excited Linear Prediction (CELP) system, it is necessary to adjust the output bit-rate according to an analysis of the immediate input speech statistics. This thesis describes a lowcomplexity variable-rate CELP speech coder for implementation on the TMS320C51 Digital Signal Processor. The system implementation is user-switchable between a fixed-rate 8 kbit/s configuration and a variable-rate configuration with a peak rate of 8 kbit/s and an average rate of 4-5 kbit/s based on a one-way conversation with 30% silence. In variable-rate mode, each speech frame is analyzed by a frame classifier in order to determine the desired coding rate. A number of techniques are considered for reducing the complexity of the CELP algorithm for implementation while minimizing speech quality degradation. In a fixed-point implementation, the limited dynamic range of the processor leads to a loss in precision and hence a loss in performance compared with a floating-point system. As a result, scaling is necessary to maintain signal precision and minimize speech quality degradation. A scaling strategy is described which offers no degradation in speech quality between the fixed-point and floating-point systems. We present results which show that the variable-rate system obtains near equivalent quality compared with an 8 kbit/s fixed-rate system and significantly better quality than a fixedrate system with the same average rate. To my parents and my fiance, with love.

9 citations

Patent
Yang Gao1
11 Mar 2004
TL;DR: In this paper, an approach for improving quality of synthesized speech is presented, where the input speech or residual is first separated into a voiced portion and a noise portion using CELP methods.
Abstract: An approach for improving quality of synthesized speech is presented. The input speech or residual is first separated into a voiced portion and a noise portion. The voice portion is coded using CELP methods. The noise portion of the input speech may be estimated at the decoder since it contains minimal voiced speech components. The separation is frequency dependent and is adaptive to the input speech. The separation may be accomplished using a lowpass/highpass filter combination. The information regarding bandwidth of the lowpass/highpass is presented to the decoder to facilitate reproduction of the noise portion of the speech.

9 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
20226
20213
20207
201915
201810
201713