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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
21 Oct 1996
TL;DR: A speech code/decode algorithm which combines MBE and LPC speech model is proposed, which can operate at 2.4 kbps with much higher quality of synthesised speech than LPC-10e and less computation complexity than CELP, VSELP and so on.
Abstract: A speech code/decode algorithm which combines MBE and LPC speech model is proposed. In this model, the spectral envelope is represented using Linear Prediction Coefficients, which are coded using Line Spectrum Frequencies (LSFs). It can operate at 2.4 kbps with much higher quality of synthesised speech than LPC-10e and less computation complexity than CELP, VSELP and so on. Therefore it is particularly attractive for VLSI implementation.
Book ChapterDOI
01 Jan 1993
TL;DR: Important advances in speech compression algorithms and the availability of efficient low cost signal processors to implement these algorithms resulted in systems which can reproduce reasonably good quality speech at bit rates as low as 4,800 bits per second.
Abstract: Important advances in speech compression algorithms and the availability of efficient low cost signal processors to implement these algorithms resulted in systems which can reproduce reasonably good quality speech at bit rates as low as 4,800 bits per second. The success of these coders has stimulated considerable interest both in the scientific research community and in industrial development centers in using the low bit rate speech coding technology for emerging real-life applications. These include the cellular telephone services, the secure telephone, mobile satellite, land mobile communications, and the multi-media applications.
Journal ArticleDOI
TL;DR: An ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform is presented.
Abstract: Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.
Patent
06 Nov 1997
TL;DR: In this paper, a random codebook is made of a fixed waveform storage section, followed by a vector rearranging unit, which shifts the vectors to positions determined to minimize the quantization distorsion using a pulse placement methodology.
Abstract: not available for EP0994462Abstract of corresponding document: EP0991054In a CELP type speech coder, the excitation is quantized by vectors from a random codebook. The random codebook is made of a fixed waveform storage section (181), followed by a vector rearranging unit (182). The rearranging section (182) shifts the vectors to positions determined to minimize the quantization distorsion using a pulse placement methodology of an algebraic coder. The vectors are summed (183) to generate the excitation code vector.
Proceedings ArticleDOI
Lin Yin1
09 Jun 1997
TL;DR: A signal dependent adaptive-switched predictor is developed that not only delivers significant coding gain for stationary signals but also recovers quickly from transient signals.
Abstract: In this paper, block backward adaptive linear predictors are used for improving the performance of perceptual audio codecs. Based on the investigation on different linear prediction algorithms, a signal dependent adaptive-switched predictor is developed. This predictor not only delivers significant coding gain for stationary signals but also recovers quickly from transient signals. At a bitrate of 64 kbit/s, the performance of the new codec for most critical test sequences is significantly better than MPEG-1 Layer II.

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