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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 split vector quantization scheme which employs interframe prediction has shown to be an attractive approach to encode the synthesis filter parameters and provides a performance comparable to the IS-54 scheme and uses 10 bits less for each LSF frame.
Abstract: Efficient quantization of synthesis filter coefficients for CELP (Code Excited Linear Prediction) coders is essential to achieve high quality speech at low rates. Three vector quantizers with good potential for utilization in low rate coders are studied. Each of them is implemented inside the structure of the VSELP (Vector Sum Excited Linear Prediction) coder, an important member in the class of CELP coders. These three vector quantizers are used to encode LSF (Line Spectral Frequency) parameters and are compared in terms of robustness to channel errors, complexity and quality of synthesized speech. Performance of synthesized speech is evaluated considering the objective measure of frequency weighted signal to noise ratio and subjective results obtained from listening tests. With the purpose of improving the robustness to channel errors, the application of simulated annealing to assign binary indices to the output levels of the quantizer is also investigated. A split vector quantization scheme which employs interframe prediction has shown to be an attractive approach to encode the synthesis filter parameters. It provides a performance comparable to the IS-54 scheme and uses 10 bits less for each LSF frame.

4 citations

Proceedings ArticleDOI
23 Mar 1992
TL;DR: A novel version of low-delay vector excitation coding (LD-VXC) operating at 8 kb/s that provides very good speech quality with a coding delay below that of the CCITT requirement and quality is comparable to that of VSELP operating at the same rate.
Abstract: The authors present a novel version of low-delay vector excitation coding (LD-VXC) operating at 8 kb/s that provides very good speech quality with a coding delay below that of the CCITT requirement. The coder integrates techniques such as vector quantization and analysis-by-synthesis, without any excessive buffering of speech samples in the encoder, by incorporating backward adaptive linear prediction. Compared with LD-VXC at 16 kb/s which was reported earlier, this version features an increased vector dimension, closed-loop pitch prediction with a three-tap pitch filter, codebook orthogonalization techniques, and dual-mode interframe coding of the pitch trajectory. Informal listening tests demonstrate that its quality is comparable to that of VSELP operating at the same rate, while maintaining a delay constraint an order of magnitude smaller than that of VSELP. >

4 citations

Journal ArticleDOI
TL;DR: From experimental results, it is shown that the proposed coder provides a synthesized speech signal that is quite comparable in quality to that of the conventional CELP coder with low computational complexity.
Abstract: The major drawback of the code excitation linear prediction (CELP) coder is computational complexity that finds the best excitation vector from a stochastic codebook. To provide a synthesized speech signal with reasonable quality, the size of the stochastic codebook should be large. For this reason, the search becomes highly complex. To overcome this difficulty, several methods have been proposed. In this paper, we consider a method that enables us to directly determine the stochastic excitation vector without a codebook search. The stochastic excitation vector has been determined by projection onto a subspace that is obtained using the Karhunen-Loeve (K-L) expansion and the spectral property of the random excitation residual vector. Since the excitation vector can be determined without a codebook search, the computational complexity becomes low. From experimental results, it is shown that the proposed coder provides a synthesized speech signal that is quite comparable in quality to that of the conventional CELP coder with low computational complexity.

4 citations

Patent
25 Sep 2007
TL;DR: In this paper, the inter time difference between the first audio signal and the second audio signal is determined by an algorithm based on determining cross correlations between first and second audio signals, which is then determined by a cross-correlations algorithm.
Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103) An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals

4 citations

Proceedings ArticleDOI
Claude Galand1, J. Menez, M. Rosso
03 Sep 1990
TL;DR: The authors propose a modification of the classical CELP (code-excited linear predictive) algorithm in order to reduce its computational complexity and required memory size, while preserving the quality of the reconstructed speech.
Abstract: The authors propose a modification of the classical CELP (code-excited linear predictive) algorithm in order to reduce its computational complexity and required memory size, while preserving the quality of the reconstructed speech. Rather than performing the individual weighting of each candidate sequence, the authors suggest a global implementation of the vocal tract weighting function at the code-book level, thanks to the use of an adaptive code-book. As a result, the analysis-by-synthesis procedure does not require the processing of all the candidate sequences through the synthesis and weighting filters, and therefore the complexity requirement of the algorithm is much reduced. The method was demonstrated on a 7.2 kb/s adaptive code-book CELP operating with the same level of quality as the 13 kb/s GSM coder which has been normalized for speech coding in the European cellular radio network. The proposed technique was then applied to the design of a coder operating with a low delay requirement. >

4 citations


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