搜索资源列表
thesis
- this the document for speech compression-this is the document for speech compression
sobi
- SOBI算法，基于二阶统计量的独立成分分析算法，用于混合语音的盲分离。-SOBI algorithm, Second-Order Blind Identification algorithm, used in blind speech signal separation
0987542
- 一个声纹识别的全代码,包括语音信号的预处理,建模,和识别-Voiceprint identification of a full code, including the speech signal pre-processing, modeling, and identification
ogrady2007_phd
- 国外欠定语音盲分离的博士论文，作者为Paul D. O’Grady，LOST算法的作者。该博士论文包括语音信号分离，非负矩阵分解等内容。-Sparse Separation of Under-Determined Speech Mixtures，A dissertation submitted for the degree of Doctor of Philosophy
F2_6764
- 端点检测是指用数字处理技术来找出语音信号中的各种段落（如音素、音节、词素、词等）的始点和终点的位置。语音段起止端点检测是语音分析、语音合成和语音识别中的一个必要环节。传统的端点检测方法是从wav文件中获取语音采样，将其分帧并计算短时能量和过零率参数，然后进行端点检测。这种工作方式被称为离线处理方法 ，无法实现语音信号的实时处理，对于语音信号分析具有一定的局限性。本文通过开发ActiveX控件，在MATLAB环境下将其嵌入到figure窗口中，以GUI程序的方式使用，实现语音信号端点检测的实时处
voicebux
- VOICEBOX: Speech Processing Toolbox for MATLAB
am_fm
- Basic familiarization about AM and FM, i have taken my speech to check whether AM or FM properly recover the signal or not, after modulation and demodulation.
dist_LLR
- Log Likelihood Ratio (LLR) Objective Speech Quality Measure
5
- 基于小波变换的语音增强_阙值去噪的研究基于小波变换的语音增强_阙值去噪的研究-Wavelet-Based Speech Enhancement _ Que denoising based on wavelet transform of the study of speech enhancement denoising _ Que Research
yuyincaiji
- 语音采集与回放系统源代码：1.为了使读音数据存储的时间更长，速度更快，选用了256K*16Bit的SRAM；2.为了减少单片机的控制复杂度，使用了FPGA来控制SRAM的读写操作，节约了不少单片机的I/O资源；3.为了以后的高速数据存储，本设计中加入了fifo,其位宽及深度可在程序中自由设置，方便灵活。-Speech acquisition and playback system source code: 1. In order to make pronunciation longer data
textToSpeech
- text to speech with microsoft speech object library.
speech
- 语音识别系统，是在vc平台编写的，仅供参考-Speech recognition system, for reference only
Speech_Test
- In this project we have processed the speech signal with the help of the DIGITAL SIGNAL PROCESSING techniques. The speech signal is given as the input will be verified using speech recognition technique using matlab. We have used Mel Frequency Cepstr
Speech_LPC
- This GUI is used to analyze the speech signal at the selected region of 256 samples. All the calculation is based on the sampling of 8 KHz. First 3 formants of the selected block of samples are derived from the LPC-8 coefficients. -This GUI is used t
GMM1_NE
- Gaussian Mixture Models (GMM) for speech noise reduction
aecfortelenetwork
- 這篇論文提出了一個在網路電話上處理回音消除的架構，並考量到雙邊通話提出了改進的演算法。-a model for electrical echo in telephone networks for varied echo-path delay was designed. This was with a view to implementing an adaptive electrical echo canceller based on a combination of the No
ho
- a tutorial for some methodes in speech enhancement
Speech
- 实现文本语音功能，将指定字符串转为语音。-text-to-speech
DigRecog
- Demonstration for Digit Recognition (and later speech recognition) using HMM Models
celp
- celp code used for speech coding using the matlab software which is used to form speech analysis