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.

Vector predictive quantization of the LPC spectral parameters for low‐rate speech coding

Abstract: Scalar quantization of the LPC parameters requires a high bit rate. Considerably lower rate can be obtained via vector quantization (VQ). However, complexity constraints dictate the use of suboptimal VQ. The use of vector predictive quantization (VPQ) for the LPC parameters is proposed. In VPQ, the suboptimality is compensated for by exploiting the temporal redundancy in the input. VPQ is a two‐stage memory VQ. In the first stage, the input vector is predicted from quantized past vectors, using a set of vector coefficients, held in a predictor codebook. In the second stage, the predicted vector is combined with a residual vector to form the final output. A set of residual vectors is held in a residual codebook. VPQ was applied to the LPC parameters in the down‐sampled, log‐magnitude spectral domain. The idea was to design an efficient VPQ under the perceptually meaningful log‐likelihood distortion measure, while circumventing the stability problem of the synthesis LPC filter. This VPQ was used in a 4.8 kbit/s CELP coder where only 1.0 kbit/s were allocated to the parameters. The performance was almost indistinguishable from that of a CELP coder with unquantized parameters. [Work supported by NSA.]

A 4.1 kb/s hybrid speech coder

Abstract: In this paper, a hybrid speech coder based on a mixture of sinusoidal and CELP coding algorithms is introduced. The conventional harmonic sinusoidal coding algorithm is revised in terms of fundamental frequency estimation, unvoiced classification and decomposition of speech spectrum into two regions of harmonic and non-harmonic components. Sinusoidal and CELP coding algorithms are used to code the harmonic and non-harmonic components, respectively. Peaks of speech spectrum are, first, interpolated and, then, represented by linear predictive all-pole filter parameters. A harmonic tracking algorithm interpolates the sinusoidal parameters between adjacent frames to introduce an essential level of periodicity in the synthetic speech waveform. Pure unvoiced frames are only coded by CELP algorithm. Spectrograms of the original speech and its synthetic versions synthesised by the proposed hybrid speech coder operating at 4.1 kb/s and the Motorola GSM half-rate speech coder operating at 5.6 kb/s are compared in both noise-free and noisy environments.

Speech coding algorithms: lpc10, adpcm, celp and vselp

Abstract: The past decade has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer- related voice applications. Central to this progress has been the development of new speech coders capable of producing high- quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. In mobile communication systems, service providers are continuously met with the challenge of accommodating more users within a limited allocated bandwidth. For this reason, manufactures and service providers are continuously in search of low bit-rate speech coders that deliver tollquality speech. The objective of this project is to compare three commonly used algorithms in wireless communication systems: ADPCM, CELP and VSELP. The project report starts with the description of these speech coders. Then we present our implementation results and finally give concluding remarks followed by comments on future research in this area.

Variable-rate residual-transform vocoders using auditory perception approximation

Abstract: In this invention, a simple variable rate vocoder for compressing the speech with low computational complexity is proposed by exploiting an adaptive pitch predictor, discrete cosine transform (DCT) on its residual signal, and perceptual representation of the DCT coefficients. To improve the speech quality, we further suggest pre-emphasized and de-emphasized filters respectively figured in the encoder and the decoder. The proposed speech coder, which achieves a reasonable quality but with much lower computational complexity than the traditional coded excited linear prediction (CELP) coders, can simultaneously offer multiple bit rates with various speech quality in the same coding structure. The real-time implementation comprising both encoder and decoder in a single low cost DSP chip confirms that the proposed vocoder produces an acceptable speech quality. With simple and variable-rate advantages, the proposed vocoder is suitable for providing digital speech communication over PSTN, internet, and wireless networks to achieve quality-of-service applications.