ToolboxBSS

自制DMX512控制台,很不错的一个软件,他对灯光的控制,非常出色.(Homemade DMX512 console, a very good one software, he lighting control, very good.)

本代码是基于matlab实现的melp算法，编码速率为2.4kb/s(This code is based on the matlab implementation melp algorithms, coding rate of 2.4kb/s)

本程序音频共有四个，借助鸡尾酒会问题思想，用ica算法实现四个音频的分离(The audio program for a total of four, with the help of the cocktail party problem, realizing the separation of four audio using ICA algorithm)

全面介绍语音合成的知识比较系统很适合语音学习(A comprehensive comparative introduction speech synthesis system is suitable for knowledge language learning)

espeak语音生成库，内附有移植使用文档《espeak移植使用》(espeak speech generation library, containing transplanted using the document espeak transplant use )

语音信号处理c语言程序-音高、加窗、MFCC、PLP、等(C programming language speech signal processing- pitch, add window, MFCC, PLP, etc)

为均衡带限信号所引起失真的横向或格型自适应均衡器(其中横向FIR系统长M=11), 系统输入是取值为±1的随机序列 ，其均值为零；参考信号 ；信道具有脉冲响应： 式中w用来控制信道的幅度失真(w = 2～4，例如，取w = 2.9,3.1,3.3,3.5等),而且信道受到均值为零、方差为 (例如，取 ，相当于信噪比为30dB)的高斯白噪声 的干扰。试比较基于下列五种算法自适应均衡器在不同信道失真、不同噪声干扰下的收敛情况(对应于每一种情况，在同一坐标下画出其学习曲线)： 横向/格-梯型结构LMS算法[1-4] 横向/格-梯型结构RLS算法[1-4] ()

根据房间配置信息生成房间传函，进而生成模拟语音数据(Depending on the room configuration generates a room transfer function, thereby generating an analog voice data)

观测实时模拟信号（语音）的频谱 用音频设备采集一段语音，将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的，应把它分段分析，而且实际运用中的数字信号处理的FFT的点数是有限的，一般只能达到千点。用傅里叶反变换IFFT，从频域恢复信号。画出频谱图和语音波形图。 (Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw the spectrum map and voice waveform.)

语音库开发工具源码，例程程序创建机读SAPI.SpVoice对象开发语音库。(Speech development tools source , routine program to create machine-readable SAPI.SpVoice speech object development .)