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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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Proceedings ArticleDOI
M. De Meuleneire, Herve Taddei1, O. de Zelicourt1, Dominique Pastor, P. Jax 
14 May 2006
TL;DR: Listening tests suggest that the proposed codec is equivalent to the ITU-T G. 722 at 48 kbit/s for speech signals, and the relevance of such a scheme when compared to a pure wavelet packet decomposition.
Abstract: This paper presents a scalable wideband speech codec working at bitrates ranging from 8 to 32 kbit/s. The core layer is the ITU-T G. 729 at 8 kbit/s. A first enhancement layer is a bandwidth extension algorithm requiring 2 kbit/s to widen the G. 729 narrow band output speech. The difference between the wideband original and reconstructed signal is transformed in the time-frequency domain by a full wavelet decomposition. The resulting coefficients are quantized by an embedded quantizer at 22 kbit/s. Listening tests show the relevance of such a scheme when compared to a pure wavelet packet decomposition. In addition, listening tests suggest that the proposed codec is equivalent to the ITU-T G. 722 at 48 kbit/s for speech signals.

8 citations

Proceedings ArticleDOI
07 Sep 1997
TL;DR: The design, implementation and performance of a high quality low bit rate speech codec for wireless communication is presented, based on the CELP model, resulting in robustness to transmission errors and high quality across changing speech levels and background noise conditions.
Abstract: The design, implementation and performance of a high quality low bit rate speech codec for wireless communication is presented. The codec is based on the CELP model. Generalized analysis-by-synthesis, algebraic fixed codebooks, and multistage LSF techniques are used, resulting in robustness to transmission errors and high quality across changing speech levels and background noise conditions. The bit allocations for the quantization of LSF, pitch and the excitation are chosen in a mode specific manner based on a robust mode classification scheme. A 4.8 kb/s version has been implemented and subjective tests show speech quality that is equivalent to or better than most cellular standard codecs. Performance is also consistent across speech levels and transmission errors.

8 citations

Journal ArticleDOI
J.I. Lee1, C.K. Un1
TL;DR: The proposed multistage SELP coder has much lower complexity than a code-excited linear prediction (CELP) coder, but yields comparable output speech quality at the same transmission rate.
Abstract: We present an approach to low-rate speech coding based on multistage self-excited linear prediction (SELP). The coder has much lower complexity than a code-excited linear prediction (CELP) coder, but yields comparable output speech quality at the same transmission rate. In the proposed multistage SELP coder, the long-term predictor is modified to model better the onset of a voiced section of speech or voice with high pitch.

8 citations

Proceedings ArticleDOI
25 Nov 2001
TL;DR: An adaptive voice playout method for handling network delay jitter in voice over packet (VOP) receivers and a novel, voicing-classification based speech extension algorithm for CELP speech coders is developed.
Abstract: We propose an adaptive voice playout method for handling network delay jitter in voice over packet (VOP) receivers. Our method allows playout delay increase during both silence periods and active speech; however, it allows playout delay decrease during silence periods only. Since the playout delay is increased during active speech, a speech extension algorithm is required. Therefore we have developed a novel, voicing-classification based speech extension algorithm for CELP speech coders. Though the complementary speech truncation algorithm is not needed for our adaptive playout mechanism, it is suitable for applications such as voice synchronization with other media.

8 citations

Journal ArticleDOI
TL;DR: A “coding gap” of roughly 32-2.4 kbit/s is shown to actually define “medium-rate” speech coding, and the fundamental approaches trying to close the gap are exposed.

8 citations


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