Demo script for testing for block

语音识别与合成入门3，word或压缩包中内有matlab源码-speech recognition and synthesis 3 entry, word or compressed packages within Matlab FOSS

WAVE文件作为多媒体中使用的声波文件格式之一，它是以RIFF格式为标准的。RIFF是英文Resource Interchange File Format的缩写，每个WAVE文件的头四个字节便是“RIFF”。WAVE文件由文件头和数据体两大部分组成。其中文件头又分为RIFF／WAV文件标识段和声音数据格式说明段两部分。WAVE文件各部分内容及格式见附表。-WAVE as a multimedia documents used in the acoustic format, it is based on RIFF format as a standard. RIFF English Resource Interchange File Format of the initials, each WAVE document for the first four bytes is "RIFF." WAVE document from the document and data on the first two major components. The first document which is divided into RIFF/logo of WAV files and voice data formats note of two parts. WAVE part of the document content and format of the attached tables.

这是微软TTS语音引擎（SAPI5.1）Delphi应用程序的详细说明。不幸的是，它是用英语写的。：（如果有的话，请帮助翻译什么。）

录音部分是参照vckbase的录音api代码,结合了一小段socket(TCP)就可以用来单方说话录音了,程序分两部分一部分是(录音机+网络发送代码),一部分是(接收数据+播放波形音乐代码),由于程序只是为了自己用,很多地方没有注意错误的识别,结构也比较乱,本程序当初最头疼的就是控制损耗内存,结果用了双缓存来存储波形数据来交替的存储/清除. -part of the recording light vckbase recording api code combination of a small socket (TCP) can be used for recording a unilateral words, part of the procedure is divided into two parts (recorders network code), part of the (receiving data waveform music broadcast code), as a result of procedures only for their own use. many places no attention to the wrong identification, the structure was chaotic, most of the first procedure is a headache for the control of memory loss, Results of the double-caching to store waveform data to the turn of the storage/removals.

ZylGSM 是一个 Delphi 和 C + + 生成器的组件，与 GSM 调制解调器或电话与集成调制解调器 （几乎所有新的一代移动电话） 进行通信。此组件与工作在 (Hayes） 很想加入书友会 GSM 调制解调器连接到串行端口之一。您可以使用它也与 USB、 IrDA、 蓝牙设备，因为这些设备有一个驱动程序，将重定向到一个虚拟串行端口从 USB、 IrDA 或蓝牙端口输入 什么是此版本中的新功能： 版本 1.28 支持 Delphi / C + + 生成器 XE3。

文本到语音，需要事先安装SAPI,可实现tts(text to speech)-Text to voice, the need for pre-installed SAPI, can realize tts (text to speech)

语音信号的频域处理,语音虽然是一个时变、非平稳的随机过程。但在短时间内可近似看作是平稳的。因此如果能从带噪语音的短时谱中估计出“纯净”语音的短时谱，即可达到语音增强的目的。由于噪声也是随机过程，因此这种估计只能建立在统计模型基础上。利用人耳感知对语音频谱分量的相位不敏感的特性，这类语音增强算法主要针对短时谱的幅度估计。 -voice signals in the frequency domain processing, voice is a time-varying, nonstationary random process. But in a short period of time can be approximated as smooth. So if Noisy Speech from the short-term spectrum estimate "pure" voice of the short-term spectrum, and reached speech enhancement purposes. As the noise is random process, which can only be estimated based on statistical models based on. Use ear perception of voice spectrum component of the phase sensitive to the characteristics of such speech enhancement algorithms targeted at the rate of short-term spectral estimation.

C54X语音压缩编码，采用ADPCM算法，- C54x speech compress codec,based on ADPCM algorithm.

用于语音识别的汉语树形词典，完成了6万词汇的建树操作及相关子节点等的查询处理，主要用于指导语音搜索整个建立时间大概为3分钟-for speech recognition dictionary of Chinese tree, completed 60,000 contribution to the vocabulary and related operations, such as sub-nodes of query processing, Pronunciation guide for the main search the entire building for about three hours minutes

matlab的语音处理工作箱，包括分帧，参数计算等等各种功能-The matlab toolbox of speech processing,including enframe, parameter calculating and so on.