《FUNDAMENTALS OF SPEECH RECOGNITION（语音识别基本原理）》（英文），大家可以好好学习下...

Hilbert_Huang变换和仿真系统设计,硕士学位论文，研究了多种时频算法，改进了HHT算法，并进行了比较。-Hilbert_Huang transformation and simulation system design, a master degree thesis, research a variety of time-frequency algorithm, improved HHT algorithm and compared.

语音识别与合成入门5，word或压缩包中内有matlab源码-speech recognition and synthesis 5 entry, word or compressed packages within Matlab FOSS

语音信号分帧加窗程序，许多程序包里都没有的！-Speech signal windowed framing procedure, many programs do not have the bag!

adpcm压缩算法，测试过，效果不错-Adpcm compressing algorithm. Has good behaviour through testing.

希尔伯特变换的C++源代码

16kbpsCVSD与64kbpsPCM编码数字转换算法描述及实现-16kbpsCVSD and 64kbpsPCM code conversion and encoding algorithm description and implementation

声音信号的短时能量matlab实现程序源码-Audio signal to achieve short-term energy program source code matlab

Linux下的CELP语言编码，压缩语音质量极好。支持2K到32K的压缩。实现部分使用C语言编写。可用做Linux下的动态库，有例子。-under Linux CELP language coding, compression excellent speech quality. Support 2 K to 32 K compression. Part of the use of C language. Linux can be done under the DLL, is an example.

G.723语音编码 The source code has been modified by simplifying the ADPCM algorithm to increase running speed. Here, the quantizer scale factor, y(k), is determined by "unlocked" factor yu(k) only. The factor yl(k) and speed-contral factor al(k) are ignored.-G.723 speech coding The source code has been modified by simplifying the ADPCM algorithm to increase running speed. Here, the Quantizer scale factor, y (k), is determined by "unlocked" factor yu (k) only. The factor yl (k) and speed-contral factor al (k) are ignored.

经TLV320AIC23音频编解码器，一方面送入DSP存储器，另一方面方便用户实现录音和播音-By TLV320AIC23 audio codec, on the one hand into the DSP memory, on the other hand, realize user-friendly recording and broadcasting