matlab中wave文件的绘图功能源代码，可以

此程序为自相关法进行基音检测的程序，主要应用于语音识别。-program for petch detection using auto-correlation method, mainly for speech recognition

字符的分割与识别在matlab环境下实现-character segmentation and recognition of the environment under Matlab

txt2wav.rar文本语音（Text-to-Speech，以下简称TTS），它的作用就是把通过TTS引擎把文本转化为语音输出。代码示范了如何运用Microsoft Speech SDK 建立自己的文本语音转换应用程序。应该事先下载微软的语音SDK里面含微软的语音开发工具原来是sapi5.1现在好像升级了微软的MSDN上有下-txt2wav.rar text pronunciation (Text-to-Speech, hereafter refers to as TTS), its function is transforms through the TTS engine the text as the pronunciation output. How did the code demonstrate has established own using Microsoft Speech SDK text pronunciation transformation application procedure. Should beforehand download Microsoft inside pronunciation SDK to contain Microsoft the pronunciation development kit originally is sapi5.1 now looks like promotes on Microsoft s MSDN to have next

给信号加入噪声时，信噪比的控制小程序，基于matlab。-to signal to noise, the signal-to-noise ratio of the control of small procedures, based on Matlab.

此程序： 5 秒记录您的声音 播放录制 在双精度数组中存储的数据 绘制波形 在该示例中的所有帧计算 LPF 900 Hz 要删除上部频率 （因为球场信息.将会低于 900 Hz） 上的讲话 执行中心裁剪和找到剪辑水平 （CL） 选择适当的修剪级别 执行中心削波 计算自相关函数 calcute 球场 计算整个样本 X 的

最新的ITU-T的宽带语音编解码标准G.729.1,是对原先的G.729的最好的调整.码流输出速率可以进行自适应调整.满足未来通信要求.希望对大家有所帮助.-The latest ITU-T

本程序可以产生一组正态模拟数，用来模拟实际生产中服从或近似服从正态分布的数据，很有用。-This procedure can generate a group of normal simulation the number used to simulate the actual production or similar to obey normal distribution of data, very useful.

nlms算法及其示例,与大家分享,希望大家可以帮我弄个多输入的NLMS-nlms algorithm and its examples to share with you, I hope everyone can help me multi-input弄个the NLMS

线性量化。读取WAV文件，用不同的比特数量化，抽样频率在8khz-Linear quantization. Read WAV files, with different number of bits, the sampling frequency of 8kHz

数字音频格式有很多种，其质量与采样频率和采样精度两个参数有关。频率的表示单位为赫兹〔Hz〕，它表示每秒采样次数。采样频率越高，音质就越好。采样精度为每次采样所存储的数据数量，它决定每个数字信号所能够表示的离散振幅的数量。存储每个样本的数据越多，音质就越好。但是高品质的声音需要占用大量的内存和磁盘空间。考虑到网络带宽，在Internet连接上传输就需要花费很长的时间。对于Applet来说，保证声音文件的最小化是极为重要的。-digital audio format there are many, the quality and frequency of sampling and sampling accuracy of the two parameters. The frequency of Hertz unit of [Hz], it said the number of samples per second. The higher sampling frequency, the sound quality better. For each sampling precision sampling by the amount of data storage, it was decided that each digital signal can be expressed as the number of discrete amplitude. Each sample storage of more data, better sound quality. But high-quality voice would take a lot of memory and disk space. Consideration of the network bandwidth, the transmission on the Internet connection on the need to spend a very long time. Applet for instance, to ensure that voices of the smallest document is very important.