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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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Book ChapterDOI
W. Bastiaan Kleijn1
01 Jan 1991
TL;DR: This paper focuses on improving source-dependent channel codes through optimization of objective error criteria by improving the robustness of the source coder and the quality of the resulting codes.
Abstract: Efforts to minimize the effect of channel errors on low-rate speech coders, such as code-excited linear prediction (CELP), can be divided into methods which change the robustness of the source coder [1], and procedures which add error correction and error detection by means of a separate channel coder. The channel coder often provides several levels of protection, with more error protection being assigned to the parameters or bits which are judged to be more sensitive to channel errors [2–4]. Such protection can be considered to be a form of source-dependent channel coding. This paper focuses on improving source-dependent channel codes through optimization of objective error criteria.

11 citations

Proceedings Article
18 Oct 2012
TL;DR: The proposed sparse autoregressive (SAR) signal representation is given in a factorized form - the model is a cascade of the so-called formant filter and pitch filter in order to perform a projection-based signal interpolation.
Abstract: This paper presents a new approach to elimination of impulsive disturbances from archive speech signals. The proposed sparse autoregressive (SAR) signal representation is given in a factorized form — the model is a cascade of the so-called formant filter and pitch filter. Such a technique has been widely used in code-excited linear prediction (CELP) systems, as it guarantees model stability. After detection of noise pulses using linear prediction, the factorized model is converted into a generic sparse form in order to perform a projection-based signal interpolation. It is shown that the proposed algorithm is able to deal favorably with speech signals with strong glottal activity, which is a serious problem for algorithms based on the classical AR modeling.

11 citations

Book ChapterDOI
01 Jan 1991
TL;DR: C coding schemes that reduce the number of required DSPs and memory will significantly reduce the overall cost of a network.
Abstract: The trend in today’s private digital networks is to integrate data with voice traffic. This offers considerable cost savings and convenience by allowing data and digitized voice to share the same transmission media, and much of the hardware and software. Code excited linear prediction (CELP) [1] is a speech coding method that has tremendous potential for providing the high quality that network voice applications demand for rates at and below 16 Kb/s. One difficulty with CELP is that it requires tremendous computational resources and memory for its codebook search. Just as there is a cost associated with the maximum bandwidth that a given network link can accommodate, there is a cost associated with the hardware necessary to support speech compression. With new telecommunications technologies promising to significantly reduce the cost of network bandwidth, the cost of the speech processing hardware may someday become the single most important consideration. Since digital signal processing chips (DSPs) and the fast memory chips supporting them consume a relatively major portion of a speech coding board’s real estate and power consumption, coding schemes that reduce the number of required DSPs and memory will significantly reduce the overall cost of a network.

11 citations

Proceedings ArticleDOI
08 Jun 1994
TL;DR: A novel high-quality, low-complexity dual-rate 4.7/6.5 kbits/s algebraic code excited linear predictive (ACELP) codec is proposed for adaptive speech communicators, which can drop their source rate and speech quality under network control in order to invoke a more error resilient modem amongst less favourable channel conditions.
Abstract: A novel high-quality, low-complexity dual-rate 4.7/6.5 kbits/s algebraic code excited linear predictive (ACELP) codec is proposed for adaptive speech communicators, which can drop their source rate and speech quality under network control in order to invoke a more error resilient modem amongst less favourable channel conditions. Source-matched binary Bose-Chaudhuri-Hoequenghem (BCH) codecs combined with unequal protection diversity- and pilot-assisted 16and 64-level quadrature amplitude modulation (18-QBM, 64-QAM) are employed In order to accommodate both the 4.7 and the 6.5 kbits/s coded speech bits at a signalling rate of 3.1 kBd. In a bandwidth of 200 kHz 32 time slots can be created, which allows to support in excess of 50 users, when employing packet reservation multiple access (PRMA). Good communications quality speech is delivered in an equivalent bandwidth of 4 kHz, if the channel signal-to-noise ratio (SNR) and signal-to-interference ratio (SIR) are in excess of about 15 and 25 dB for the lower and higher speech quality 16-QAM and 64-QAM systems, respectively. >

11 citations

Patent
16 Jun 2005
TL;DR: In this paper, an audio encoding device capable of realizing effective encoding while using audio encoding of the CELP method in an extended layer when hierarchically encoding an audio signal was disclosed.
Abstract: There is disclosed an audio encoding device capable of realizing effective encoding while using audio encoding of the CELP method in an extended layer when hierarchically encoding an audio signal. In this device, a first encoding unit (115) subjects an input signal (S11) to audio encoding processing of the CELP method and outputs the obtained first encoded information (S12) to a parameter decoding unit (120). The parameter decoding unit (120) acquires a first quantization LSP code (L1), a first adaptive sound source lag code (A1), and the like from the first encoded information (S12), obtains a first parameter group (S13) from these codes, and outputs it to a second encoding unit (130). The second encoding unit (130) subjects the input signal (S11) to a second encoding processing by using the first parameter group (S13) and obtains second encoded information (S14). A multiplexing unit (154) multiplexes the first encoded information (S12) with the second encoded information (S14) and outputs them via a transmission path N to a decoding device (150).

11 citations


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