【作者】 王都生；

【导师】 樊昌信；

【作者基本信息】 西安电子科技大学 ， 通信与电子系统， 1998， 博士

【摘要】 本文综述了语音编码发展的现状，全面系统地分析、介绍了多带激励与混合激励线性预测语音编码的原理，重点研究了多带激励与混合激励线性预测低速率语音编码的实现方法。 在多带激励低速率语音编码研究中，本文依据动态规划技术原理，提出了一种新的、有效而实用的前后各看两帧的双向基音跟踪平滑算法。实验结果表明，采用该方法对基音周期进行跟踪平滑处理，可去除语音帧之间基音周期的“跳跃”点，保证基音周期轨迹的正确性，显著提高基音周期估计的准确度。并设计了两级离散余弦变换的编码方案，采用该方案对逐帧变化的谱幅度参数进行自适应动态量化编码，大大减少了编码比特数，实现了2.4kb/s及其以下速率的多带激励语音编码器。最后，设计了完整的2.4/1.2kb/s多带激励语音压缩编码方案，并采用数字信号处理器TMS320C31实时实现了该方案，开发了一个实用的声码器产品。目前该声码器已经应用于许多通信系统进行语音保密通信或扩大系统通信容量。通信实验及客观测试结果表明，两种速率的编码方案均达到实用化要求。 在混合激励线性预测低速率语音编码研究中，提出了一种新的归一化自相关函数实时基音检测算法。在算法中采用了整数基音周期估计、分数基音周期精估及基音周期倍数检验等处理步骤。实验结果表明，本文的新方法可在当前帧内实时地提取精确的基音周期估计值。设计了一种新的线谱频率参数多级矢量量化编码方案。主观和客观评价结果表明，以每帧25比特对线谱频率参数进行编码，该量化器可提供透明的量化质量。最后，设计了完整的2.4kb/s混合激励线性预测语音编码方案。在该方案中，引入了混合激励源、非周期脉冲、自适应谱增强、脉冲散布及付氏幅度模型五个附加特征来进一步改善合成语音的质量。计算机模拟及非正式的主观试听结果表明，由该方案得到的合成语音质量接近于4.8kb/s码激励线性预测语音编码算法的质量，优于本文2.4kb/s多带激励语音编码方案的语音质量。更多还原

【Abstract】 AbstractThe current situation of speech coding is summarized in this paper. The principles of the multi-band excitation(MBE) and mixed excitation linear prediction(MELP) speech coding are analyzed and introduced in detail with emphasis on the implement method of MBE and MELP speech coding at low bit rate.In the study of MBE speech coding, a new efficient practical look-back and look-ahead pitch tracking smoothing algorithm is presented based on the dynamic programming approach. The experimental results show that it can remove the umper?of pitch between the frames, guarantee the exactness of pitch track and improve the accuracy of pitch estimation. The two stage discrete cosine transformation coding scheme is designed. Using this scheme spectral magnitude parameters changing frame-by-frame are quantized adaptively and the coded bit rate of speech is largely reduced. Finally, a 2.4/1.2kb/s MBE speech coder is designed completely and implemented with the digital signal processor TMS32OC3 1. Recently our vocoder has been applied in many communication systems. The application result and DRT test show that the 2.4/1.2kb/s MBE vocoder achieves the practical requirement already.In the research of mixed excitation linear prediction speech coding, a new real-time normalized correlation function pitch detection algorithm is proposed. The integer pitch calculation, the fractional pitch refinement, and the pitch doubling check procedure are applied in the algorithm. The experimental results show that the proposed algorithm can obtain the exact pitch value within the current frame. A new multi-stage vector quantizer(MSVQ) of line spectral frequency parameters is designed. The objective and subjective evaluations show that the proposed scheme offers transparent quantization quality with 25b/frame. Finally, a 2.4kb/s MELP speech coder is designed completely. It contains five additional features. These are: mixed excitation, aperiodic pulses, adaptive spectral enhancement, pulse dispersion, and Fourier magnitude modeling. These features allow MELP speech coder to mimic more of the characteristics of natural human speech. The computer simulation and informal listening test show that the reconstructed speech quality with this coder at 2.4kb/s is closed to CELP algorithm at 4.8kb/s, and superior to 2.4kb/s MBE algorithm proposed in this paper.更多还原