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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.


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Patent
28 Dec 2000
TL;DR: In this article, a method for improving process time and speech quality of G.723.1 and reducing bit rate in a CLEP (Code Excited Linear Prediction) voice coder (or, called as vocoder) is presented.
Abstract: A method for improving process time and speech quality of G.723.1 and reducing bit rate in a CLEP (Code Excited Linear Prediction) voice coder (or, called as vocoder) includes: a method of searching MP-MLQ fixed codebook through bit predetermination includes the steps of generating a target vector with amplitude, reducing time to search an optimal pulse array through the bit predetermination and searching all of pulses if two errors have an identical value; a formant post-filtering method of extracting a reflection coefficient of a slope compensation filter to apply a multi-degree slope compensation thereto; a pitch post-filtering method including an energy level standardization step and a step of generating a signal approximate to an average energy level; a VAD algorithm method using an energy, a pitch gain and a LSP distance; and a method of enhancing a processing time of G.723.1, improving speech quality and reducing a bit rate by using a determination logic algorithm in setting a SID frame for the voice inactive interval, and a CELP vocoder using one of the methods.

30 citations

Proceedings ArticleDOI
13 Oct 2002
TL;DR: New software extensions on J-DSP to accommodate on-line laboratories for speech processing, image processing, and communications systems are described and Statistical and qualitative evaluations that assess the learning experiences of the students that use J- DSP are presented.
Abstract: J-DSP is a Java-based object-oriented programming environment that was developed at Arizona State University for use in the undergraduate DSP class [A Sapnias et al, June 2000] In this paper, we describe innovative software extensions on J-DSP to accommodate on-line laboratories for speech processing, image processing, and communications systems Significant modifications in the object-oriented GUI of J-DSP that enable simulation of feedback systems are also presented The speech processing functions enable on-line simulations of speech coding algorithms and include PCM and ADPCM quantization as well as more elaborate algorithms such as the LPC and the CELP Image processing functionalities include development of 2-D signal processing capabilities including 2-D-FFT, 2-D-filter design, and 2-D graphics and picture processing Communications functionality covers several aspects of analog and digital modulation and demodulation On-line laboratory exercises have been developed in the aforementioned areas and posted on a web site (http://jdspasuedu) This site also includes on-line evaluation forms for the exercises Statistical and qualitative evaluations that assess the learning experiences of the students that use J-DSP are presented

30 citations

Journal ArticleDOI
TL;DR: Simulation results show that higher SEGSNR and lower computation complexity can be achieved, and the pitch contour of the synthesized speech is smoother than that produced by conventional CELP coders.
Abstract: This correspondence proposes a new CELP coding method which embeds speech classification in adaptive codebook search. This approach can retain the synthesized speech quality at bit-rates below 4 kb/s. A pitch analyzer is designed to classify each frame by its periodicity, and with a finite-state machine, one of four states is determined. Then the adaptive codebook search scheme is switched according to the state. Simulation results show that higher SEGSNR and lower computation complexity can be achieved, and the pitch contour of the synthesized speech is smoother than that produced by conventional CELP coders. >

30 citations

Proceedings ArticleDOI
15 Mar 1999
TL;DR: An adaptive multi-rate (AMR) speech coder designed to operate under the GSM digital cellular full rate and half rate channels and to maintain high quality in the presence of highly varying background noise and channel conditions is developed.
Abstract: We have developed an adaptive multi-rate (AMR) speech coder designed to operate under the GSM digital cellular full rate (22.8 kb/s) and half rate (11.4 kb/s) channels and to maintain high quality in the presence of highly varying background noise and channel conditions. Within each total rate, several codec modes with different source/channel bit rate allocations are used. The speech coders in each codec mode are based on the CELP algorithm operating at rates ranging from 11.85 kb/s down to 5.15 kb/s, where the lowest rate coder is a source controlled multi-modal speech coder. The decoders monitor the channel quality at both ends of the wireless link using the soft values for the received bits and assist the base station in selecting the codec mode that is appropriate for a given channel condition. The coder was submitted to the GSM AMR standardization competition and met the qualification requirements in an independent formal MOS test.

30 citations


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