Research On CS-ACELP Speech Coding Algorithm And Implementation

This paper describes the 8kbit/s speech coding algorithm which has been standardized by ITU-T in 1996. The algorithm is based on a Conjugate-Structure Algebraic Code Excited Linear Prediction (CS-ACELP) coding technique and uses 1 Oms (80 samples at an 8 kHz sample rate) speech frames. This coder will be used for the Future Public Land Mobile Telecommunication System and will be suitable for Personal Communication Service. The coder delivers toll-quality speech (equivalent to 32kbit/s ADPCM) for most operating conditions. The coder operates on speech frames of 10ms, computes the long-term predictor coefficients, and operates in an analysis-by-synthesis loop to find the excitation vector that minimizes the perceptually weighted error signal.In this paper, the coder structure is described, the algorithm about CS-ACELP is discussed, and its central aspects are analyzed in detail. To achieve high-quality speech and real-time implementation, CS-ACELP has been revised by novel schemes. Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech. LSP parameters are quantized by multi-stage VQ with fourth-order interframe MA prediction. This scheme has little spectrum distortion, even if the two types of speech have many variations of LSP parameters. Moreover, computational complexity for implementation is reduced in adaptive and fixed-shape codebooks without degrading the quality. Multi-stage selection is adopted in the adaptive codebook; this selection uses atruncated impulse response. Improved pre-selection is proposed in the fixed-shape codebook. Subjective testing indicates that the quality of CS-ACELP is equivalent to that of the 32kbit/s Adaptive Differential Pulse Code Modulation (ADPCM) under error-free conditions and it outperforms G.726 under error condition.In this paper, Standard C is adopted in realization of the algorithm, presents program strategies and steps of algorithm of each module. The coder and decoder is tested by utterances with noise. The results are satisfying. Moreover, the paper studies the quantization of LSP and tries some other means to quantize the LSP parameter. We adopt the means as follows: first, we change the splitting dimension of the second grade codebook which was splirted into double five dimension. By testing, we find a better splitting way. Second, we optimize the codebook and choice a part of the codeword which is used most efficiently. The result is not degraded too much while the complexity is reduced. At the end of the paper the development prospect of CS-ACELP and speech coding are described.