搜索资源列表
FIRDsgn
- 有限脉冲响应过滤器。This program designs a Finite Impulse Response (FIR) filter. The window-based method is used to obtain a low-pass, high-pass, band-pass or band-stop FIR filter-finite impulse response filters. This program designs a Finite Impulse Response
col
- 一个什么都能做的语音处理软件，Manual segmentation of speech waveforms - creates label files which can be used to train speech recognition systems Waveform editing - cutting, copying or pasting speech segments Formant analysis - displays formant tracks of F1,
PPT-SoftwareEngineering
- 语音识别系统中前端PLP参数的提取和处理 孤立词识别系统中几种滤波器组的比较研究 基于Matlab命令字识别系统及实现 基于HTK的命令字识别系统及实现 基于HTK的数字串识别系统及实xian-Speech Recognition System front-end PLP parameters for the extraction and processing isolated word recognition sy
Adaptive-noise-canceller
- 自适应线性预测器，采用FIR滤波器实现。-adaptive linear predictor, the use of FIR filters achieve.
DesignLyonFilters
- Design the cascade of second order filters and the front filters (outer/middle and compensator) needed for Lyon s Passive Short Wave (Second Order Sections) cochlear model.-the cascade of second order filters and the front filters (outer / middl
lpcbwexp
- it is a open algorithem that calculate the coefficients of the lpcc filters. its efficiency could be quite different for different applications.
iirsinglefnoise
- 首先提取语音信号，并对其加单频噪声，设计IIR滤波器，滤除单频噪声，显示滤除前后时域波形的对比，并播放滤除后的语音信号。-First, extract the speech signal, and add single-frequency noise, design IIR filters to filter out single-frequency noise. Display the time-domain waveform before and after filter, and pl
fxlms
- ％％ Active Noise Control Using a Filtered-X LMS FIR Adaptive Filter ％ This demonstration illustrates the application of adaptive filters to the ％ attenuation of acoustic noise via active noise control. - Active Noise Control Using a Filtered-X L
noise_cancelled_GUI
- 使用matlab 编写的具有调整滤波器的参数来处理输入的语音信号使之具有消除噪声的功能，包括FIR和IIR两种滤波器，使用GUI界面。-Prepared with the use of matlab to adjust filter parameters to deal with input voice signal so that they are the elimination of noise features, including two types of FIR and IIR fil
exp5
- FIR 滤波器是在数字信号处理(DSP)中经常使用的两种基本的滤波器之一,另一个为IIR滤波器. -FIR filters in digital signal processing (DSP) often used in one of two basic filters, and the other for the IIR filter.
coleawin_Matlab_Speech_Analysis
- 波形和频谱双显示 记录讲话直接进入MATLAB 手动分割讲话波形-创建标签文件 波形编辑-切割，复制或粘贴 共振峰分析-显示共振轨道的F1 ， F2和F3 基音分析 过滤工具-语音信号滤波器截止频率 比较工具-比较两个波形的频谱距离使用几种措施 增加噪声-Dual time-waveform and spectrogram displays Records speech directly into MATLAB NEW Displays time-a
impulse333
- this program uses an impulse train, takes formant frequencies and filters them. output signal is synthesized sound. commented part ,controls time=d, sampling frequency=fs normal frequency=f and bits=n, then synhesizes the output sound
analoge_high_pass
- In oreder to compare Impulse Invariance method charactristics for transformation analogue filters to digital ones in highpass and bandpass and lowpass formats,here are 6 M-files that simulate a third order butterworth filter analyizing impulse respon
WienerFilters
- All three kinds of Wiener Filters, FIR, Causal, Noncausal
Blockand Sub band Adaptive Filters
- 块和子带自适应滤波器matlab源文件，内有详细说明及运行结果图-Block and Subband Adaptive Filters
project_matlab
- Levison-Durbin 语音信号处理中的线性预测编码LPC 理论、格型滤波器以及求解现行预 测方程的算法，可以实现对语音信号重要元素的分析、合成甚至识别。 基于现有的实验平台，我们可以利用 Matlab 函数来获得几个固定语音元素（如元音） 的模型系数，LPC 得到的系数组成 IIR 滤波器。利用冲击脉冲 序列作为输入，我们就可以得到原来的语音。这是一种简单的语音合成功能。-Levison-Durbin speech signal processing in li
SPC
- 信号处理源码，包含常用的各种滤波器，功率谱计算，Z变换，线性卷积等，C语言实现。-Signal processing source code that contains a variety of commonly used filters, power spectrum calculation, Z transforms, linear convolution and so on, C language.
SPF
- 信号处理源码，包含常用的各种滤波器，功率谱计算，Z变换，线性卷积等，FORTRAN语言实现。-Signal processing source code that contains a variety of commonly used filters, power spectrum calculation, Z transforms, linear convolution and so on, FORTRAN language.
SPM
- 信号处理源码，包含常用的各种滤波器，功率谱计算，Z变换，线性卷积等，Matlab实现。-Signal processing source code that contains a variety of commonly used filters, power spectrum calculation, Z transforms, linear convolution and so on, Matlab implementation.
chengxu
- 给音乐添加噪声再通过低通滤波器滤掉噪声，性能改善过后能明显看到效果。-Add noise to the music and through a low pass filter filters noise, performance improvement can be evident after the effects.