scispace - formally typeset
Search or ask a question
Topic

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
More filters
Proceedings ArticleDOI
07 May 1996
TL;DR: A split-band encoding scheme for 16 kbit/s wideband speech coding (50-7000 Hz), using 2 unequal subbands from 0-6 kHz and from 6-7 kHz, which was motivated by an experimental evaluation of the signal bandwidth of speech frames.
Abstract: We propose a split-band encoding scheme for 16 kbit/s wideband speech coding (50-7000 Hz), using 2 unequal subbands from 0-6 kHz and from 6-7 kHz. This approach was motivated by an experimental evaluation of the signal bandwidth of speech frames. The higher subband is simply represented by white noise with adjustment of the short term energy. For the lower subband code-excited linear prediction (CELP) is used. The analysis filter bank, which performs the unequal band splitting combined with critical subsampling of the sub-bands, is described. A bit error concealment technique and the bit allocation is also presented. By informal listening tests the speech quality was rated higher than the speech quality of the CCITT G.722 wideband codec operating at 48 kbit/s.

31 citations

Journal ArticleDOI
TL;DR: A new coding scheme termed the multiband LPC (MB-LPC) vocoder is proposed, which produces good and natural quality speech at 2.4 kbit/s, which is very close to MBE coded speech at 4.8 k bit/s.
Abstract: Until recently good quality digital speech transmission was not possible below about 9.6 kbit/s. With the development of code excited linear prediction (CELP) and multiband excitation (MBE) vocoders,2 high quality digital speech transmission became possible at bit rates as low as 4.8 kbit/s. Below 4.8 kbit/s however, vocoders such as LPC-103 and the channel vocoder4 which can only produce synthetic and unnatural speech are still in use. A new coding scheme termed the multiband LPC (MB-LPC) vocoder is proposed, which produces good and natural quality speech at 2.4 kbit/s. Subjective performance of speech at 2.4kbit/s produced by the CELP is very close to MBE coded speech at 4.8 kbit/s. Informal listening tests have shown that in most cases people could not tell the difference between the new 2.4kbit/s MB-LPC coder and the 4.8 kbit/s MBE vocoder.

31 citations

Patent
09 Jan 2004
TL;DR: In this article, the authors propose a method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a binary representation of the data using perceptual weighting that uses tuned weighting factors to produce a higher quality decoded voice signal than a comparable tandem transcoding solution.
Abstract: A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution. The method includes pre-computing weighting factors for a perceptual weighting filter optimized to a specific source and destination codec pair, pre-configuring the transcoding strategies, mapping CELP parameters in the CELP parameter space according to the selected coding strategy, performing Linear Prediction analysis if specified by the transcoding strategy, perceptually weighting the speech using with tuned weighting factors, and searching for adaptive codebook and fixed-codebook parameters to obtain a quantized set of destination codec parameters.

31 citations

PatentDOI
TL;DR: A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in termsof the encoding units, whereby explosive and fricative consonants can be impeccably reproduced.
Abstract: A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in terms of the encoding units, whereby explosive and fricative consonants can be impeccably reproduced, while there is an attenuation of the occurrence of foreign sounds being generated at a transient portion between voiced (V) and unvoiced (UV) portions, so that the speech with high clarity devoid of “stuffed” feeling may be produced. The encoding apparatus includes a first encoding unit for finding residuals of linear predictive coding (LPC) of an input speech signal for performing harmonic coding and a second encoding unit for encoding the input speech signal by waveform coding. The first encoding unit and the second encoding unit are used for encoding a voiced (V) portion and an unvoiced (UV) portion of the input signal, respectively. Code excited linear prediction (CELP) encoding employing vector quantization by a closed loop search of an optimum vector using an analysis-by-synthesis method is used for the second encoding unit. A corresponding decoding method and apparatus is also provided.

31 citations

Proceedings ArticleDOI
27 Apr 1993
TL;DR: A high-quality 8-bit/s speech coder based on CS:CELP (conjugate structure code excited linear prediction) with 10 ms frame length is presented and it is found that the proposed coder is robust against random bit errors.
Abstract: A high-quality 8-bit/s speech coder based on CS:CELP (conjugate structure code excited linear prediction) with 10 ms frame length is presented. To provide high quality in both error-free and error conditions, it uses four schemes: LSP (line spectrum pair) quantization using interframe correlation, preselection of codebook search, a conjugate structure, and backward adaptation of the VQ (vector quantization) gain. LSP parameters are quantized by multistage VQ with MA prediction. The preselection of the codebook reduces computational complexity and improves robustness. The CS improves the ability to handle random bit errors and reduces memory requirements. The backward adaptation of the VQ gain provides high quality and robustness without having to transmit input speech power information. Subjective testing indicates that the quality of the proposed coder is equivalent to that of the 32 kbit/s ADPCM (adaptive differential pulse code modulation) under error-free conditions. It is also found that the proposed coder is robust against random bit errors. >

31 citations


Network Information
Related Topics (5)
Decoding methods
65.7K papers, 900K citations
83% related
Data compression
43.6K papers, 756.5K citations
83% related
Signal processing
73.4K papers, 983.5K citations
83% related
Feature vector
48.8K papers, 954.4K citations
80% related
Feature extraction
111.8K papers, 2.1M citations
79% related
Performance
Metrics
No. of papers in the topic in previous years
YearPapers
20226
20213
20207
201915
201810
201713