基于MATLAB的语音共振峰的估计

题目：基于MATLAB的语音共振峰的估计

英文题目：MATLAB—BASED ESTIMATES OF

FORMANTS

院系：电子工程学院

专业：通信工程

姓名：

年级：二零零六级

指导教师：***

二零零九年十二月

摘要

语音是人类相互之间进行交流时使用最多最自然最基本也是最重要的信息载体在高信息化得今天，语音处理的一系列技术及其应用几经成为信息社会部可或缺的最重要组成部分。

语音编码算法利用语音信号的冗余信息及某些人耳不敏感的信息，可以在低比特率上获得较高质量的重建语音，压缩编码一直是通信中的关键技术。语音信号研究者们一直在寻求一种在保持语音质量不显著下降的情况下使语音信号的编码比特率最小的方法，特别地，低比特率语音编码体制(比特率在4.8 kb／s以下)因其广泛的需求而得到研究者的重视。

语音编码器的性能常常用比特率、延时、复杂度和质量4个属性来进行衡量，因此，在分析语音编码器的性能时，主要应该考虑这些属性。值得注意的是，这些属性之间不是孤立的，而是相互紧密联系的，例如，低比特率的编码器一般比高比特率的编码器有更大的延时、更高的算法复杂度和较低的语音质量。因此在对各种编码算法进行取舍时，应根据实际应用环境，在这些属性之间进行权衡。

共振峰参数编码算法在低码率的音频编码中应用越来越广泛。与基于时域波形的压缩算法相比，他在传输的过程中只需要

传输构造信号所用的基频和共振峰参数，因此可以大大地降低传输的码率，实现低码率下的多媒体通信。而且，基于共振峰参数的算法无须严格限制信号的结构，他可以灵活地描述音频信号的特征。这一灵活性决定了基于共振峰参数的算法，可以满足对音频信号进行方便访问和控制的需要。

关键字：共振峰线性预测

ABSTRACT

The human voice to communicate with each other using the most natural and the most fundamental and most important information carriers in high-information that today, a series of voice processing technology and its application several times in the information society available to the Department or the lack of the most important component of the . Speech coding algorithm using speech signals redundant information and some people's ears are not sensitive information, you can gain at low bit-rate reconstruction of a high-quality voice compression coding has been the communication of key technologies. Speech Signal researchers have been looking for a way to maintain voice quality in asignificant decline in the case of voice signals in the smallest bit rate coding methods, in particular, the low bit rate speech coding system (bit-rate of 4.8 kb / s or less) because of its wide range of demands to be researchers attention.

The performance of speech coding devices often use

bit rate, delay, complexity and quality of the four attributes to measure, therefore, in analyzing the performance of speech coder, the main consideration should be given these attributes. It is noteworthy that among these attributes are not isolated, but closely interrelated, for example, low bit rate encoder in general than high bit-rate encoder greater delay, higher algorithm complexity and the lower voice quality. Therefore, various coding algorithms to choose should be based on the actual application environment, in the trade-off between these attributes.

Formant parameter coding algorithm at low bit-rate audio coding more and more widely applied. Time-domain waveform based on the compression algorithm, the process of transmission, he need only transmit the signal structure used for the base frequency and formant parameters, it can greatly reduce the transmission rate, low bit-rate multimedia communication. Moreover, the algorithm based on formant parameters do not strictly limit the signal structure, he has the flexibility to describe the audio signal characteristics. This flexibility determines the parameters of formant-based algorithms, to meet the audio