MELP-based (Mixed Excitation Linear Prediction-based) multi-frame joint quantization low-rate speech coding and decoding method

Abstract

The invention relates to an MELP-based multi-frame joint quantization low-rate speech coding and decoding method comprising the steps that an coding end first processes signals by adopting the length of 25ms per frame, parameters, i.e. line spectrum pair frequency (lsf), pitch period (pitch), band pass unvoicing / voicing (vp) and gain (G), are sequentially extracted, joint quantization is then carried out with each three neighboring frames as a unit, wherein a three-dimensional single-code book is adopted to quantize the vp, a code book, the size of which is dynamic according to signal unvoicing / voicing, is adopted to quantize the pitch after the pitch is logarithmizd, the G is first deequalized and then quantized by a single-code book, four-stage residual vector quantization is only carried out on a 20-dimenstional vector composed of the lsf of the first frame and the lsf of the final frame, a decoding end first adopts a decoding book to process the G, the lsf and the pitch, and then adopts interpolation factor-weighted interframe linear prediction by aiming at the lsf to obtain the lsf of the middle frame, the unvoicing / voicing information of five neighboring frames of signals is embedded into the interpolation factor r solution process, and the frequency spectrum continuity and stability of the voice signals are taken into full consideration. Consequently, the method can effectively decrease the coding rate to be lower than 1.2kb / s, and has great reference value for the research and application of the low-rate speech coding technology.

Description

technical field [0001] This method relates to a low-rate speech coding method in a wireless communication system, in particular to a multi-frame joint quantization low-rate speech coding method based on mixed excitation linear predictive coding (MELP), which is suitable for wireless communication systems in communication In an environment with poor conditions and complex background noise, it occupies very few spectrum resources to realize reliable transmission of voice signals, and belongs to the field of wireless communication technology. technical background [0002] With the continuous expansion of current wireless communication services and the continuous increase in the amount of transmitted data, future wireless communication systems require higher data transmission efficiency and transmission accuracy, especially the most basic daily voice communication. However, the current wireless communication spectrum resources are becoming increasingly tight, the electromagnetic...

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Problems solved by technology

The papers "A VARIBLE LOW BIT RATESPEECH CODER BASED ON MELP" and "A 600 BPS MELP VOCODER FOR USE ON HF CHANNELS" mentioned that joint quantization of four frames and six frames is used for speech signals respectively, and the document "A MELP-based In 600b / s Very Low Rate Speech Coding, three-frame joint quantization is adopted for the speech signal at the encoding end, but only the first frame and the last frame are passed when the parameters are passed, and the inter-frame linear interpolation of the hard decision nature is adopted for the parameters at the decoding end It is predicted that the paper "Joint Optimization Algorithm of Multi-parameter Codebook Size Based on Superframe Mode" adopts dynamic codebook quantization based on signal voicing for all multi-frame parameters, which reduces the encoding rate to a certain extent, but the standard voice Subjective and objective test results show that the combination of too many single-frame parameters leads to too many dimensions of the transmitted parameters, and the time required for vector quantization in the parameter quantization process is longer, and the effect on the delay effect of the coding scheme is poor; for multi-frame parameters Too simple inter-frame linear prediction under hard decision at the decoding end leads to large distortion of parameter reconstruction, which can easily lead to interference of multiplied signals, and poor intelligibility of reconstructed signals, which is different from parameter coding to reconstruct signals. Intelligibility is inconsistent with the first goal, and all parameters of the multi-frame signal are quantized based on the dynamic codebook size of the voicing situation, which leads to the need to prepare a large number of codebooks at the encoding end when quantizing the parameters. In the specific implementation process The medium occupies a large amount of storage, resulting in low execution efficiency of parameter quantization

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