【作者】 白燕宁；

【导师】 鲍长春；

【作者基本信息】 北京工业大学 ， 信号与信息处理， 2005， 硕士

【摘要】 语音编码是实现语音数字化传输和存储的基础。拥有长途话音质量的4kb/s及4kb/s 以下速率的语音编码器是当前研究的热点。本文基于码激励线性预测语音编码模型,以散布脉冲码激励线性预测语音编码器为原型,提出了一种高质量的4kb/s 语音编码算法。该算法的固定码书采用非均匀和部分搜索域代数码书搜索算法,非均匀代数码书由代数码书的脉冲非均匀统计特性确定,部分搜索域代数码书则由代数码书矢量的周期性确定,该方法有效地弥补了低比特率情况下代数码书中脉冲数不足的缺点。在使用上述两项技术时,为保持基音的连续性,该编码器对浊音段和清音段采用了不同的基音检测方法。线谱频率参数的量化采用以加权欧式距离为测度的快速算法,前四维矢量和后六维矢量的计算复杂度分别下降为原来的23%和43%。主观和客观的听力测试表明,与原算法相比,本文所提出的4kb/s 语音编码算法的重建语音质量得到了明显改善,尤其是对女性讲话者。最后,本文在TMS320VC5410 定点DSP 开发系统上对改进的4kb/s 语音编码算法进行了实时实现。整个系统占用程序存储空间10.25 K,占用数据存储空间20.75K;编码算法复杂度约为49MIPS,解码复杂度约为5.2MIPS。客观测试结果显示,DSP 定点程序处理后的合成语音质量非常接近于浮点C 语言处理后的结果,达到了预期的要求。DP-CELP 语音编解码算法在定点DSP 上的实时实现,产生出了性价比极高的语音处理系统,在数字存储、多媒体通信等领域有着广泛的应用前景。更多还原

【Abstract】 Speech coding is the fundation of digital transmission and storage of speech. Recently,there is great interests in developing toll-quality speech coders at rates of 4kb/s and below. A high quality 4kb/s speech coding algorithm was proposed in this thesis, which is based on Code Excited Linear Prediction model and Dispersed Pulse CELP algorithm. The non-uniform and part-searching-area algebraic codebook searching algorithm is adopted in the fixed codebook. The non-uniform algebraic codebook is determined by the non-uniform statistical properties of the algebraic codebook, and the part-searching-area is determined by the periodicity of the algebraic codebook excitation vector, which makes up for the insufficient numbers of signed pulses in algebraic codebook at low bit rate. In order to preserve the continuity of pitch, different pitch detection methods are employed for voiced/unvoiced frames when these two techniques are used. The fast-search algorithm based on the weighted Euclidean distance was employed in the quantization of line spectrum frequency parameters and the complexities for calculating the the first four LSF parameters and the last six one are reduced to 23% and 43% respectively. Subjective and objective test results indicate that the reconstructed speech qualities of 4kb/s DP-CELP speech coder are improved based on these techniques, especially for the female speakers. Finally, the improved 4kb/s speech coding algorithm is implemented on the TMS320VC5410 fixed-point DSP board. About 10.25K program spaces and 20.75K data spaces are consumed by this system; about 49MIPS are needed by coder while about 5.2 MIPS by decoder. Objective test indicated there is no perceptual difference for the reconstructed speech quality compared with the original C program result. This speech processing system has a high ratio of performance to price, which is expected to be widely exploited in applications such as digital storage, multimedia communications.更多还原