搜索资源列表
dsp1.1
- LMS算法，从宽带信号中提取单频信号的算法，方法简单易行-LMS algorithm, broadband signal from the single-frequency signal extraction algorithm is simple
yycl
- 提取语音信号的lpc参数并进行时间归整，需要将wav语音文件放在指定目录下‘e:\\yyzl’-voice signal from the lpc parameters and time consolidation, need to wav sound files on the specified directory 'e : \\ yyzl'
yuyinshibiedsp
- 用DSP实现的一个简单的语音识别系统,只要实现单个词的识别即可,采样率8k,帧长30ms,帧移10ms,系统采样后分帧--端点检测,将检测到的原始语音信号保存下来,基本上一个字在30帧左右,然后提取每帧的LPC参数--将LPC参数转换为LPC倒谱系数,然后利用DTW方法和模板比较.-a brief speech recognition system, as long as the realization of a single word can be identified. 8 k sampli
pitch_detect
- 语音信号的基音提取对于语音编码，识别都十分重要，该程序用于提取语音信号的基音-voice signal from the pitch for speech coding, identification is important, The procedure used to extract the speech signal Pitch
窄带信号提取
- 语音识别中的窄带信号提取函数，matlab版-Speech Recognition narrowband signal extraction function, Matlab version
LPC.ZIP 用线性预测的方法提取语音信号中的共振峰
- 用线性预测的方法提取语音信号中的共振峰，并进行共振峰轨迹的跟踪。-Extract formants from speech,and track them.
SpeechSignalFeatureExtractioncprogram
- 主要描述了基于特定人语音信号特征提取以及相关代码-Mainly describes the people on the basis of specific speech signal feature extraction and related code
speechcode1
- matlab中用于输入语音信号，特征提取，建立模型，识别判决等三部分的源码。-matlab input speech signal is used, feature extraction, modeling, identification judgments, such as the source of three parts.
DTWspeech
- 本 文 首先 介绍了语音识别的研究和发展状况，然后循着语音识别系统的 处理过程，介绍了语音识别的各个步骤，并对每个步骤可用的几种方法在实 验基础上进行了分析对比。研究了语音信号的预处理和特征参数提取，包括 语音信号的数字化、分帧加窗、预加重滤波、端点检测及时域特征向量和变 换域特征向量.其中端点检测采用双门限法.通过实验比对特征参数的选取， 采用12阶线性预测倒谱系数作为识别参数。详细分析了特定人孤立词识别算 法，选定动态时间弯折为识别算法，并重点介绍其设计实现。 在
mfcc
- 语音信号的初始化及MFCC特征提取算法，附带测试用语音信号-Voice signal and the initialization MFCC feature extraction algorithm, with test speech signal
ACF
- 语音信号处理中基音周期提取算法。运用传统的自相关方法。并加入了三电平削波的处理。-Speech Signal Processing pitch extraction algorithm. Using the traditional autocorrelation method. And joined the three-level clipping processing.
work
- 基于负熵的FASTICA的不动点算法，并行提取信号，独立分量分析-Based on negative entropy FASTICA the fixed point algorithm, parallel extract the signal, independent component analysis
1
- 语音信号处理的课件，包含语音信号的短时分析、特征提取、矢量量化，语音编码、合成与识别-Speech Signal Processing courseware, including short-time speech signal analysis, feature extraction, vector quantization, speech coding, synthesis and recognition, etc.
EWT
- 经验小波变换结合EMD的自适应性和小波分析的理论框架,Gilles提出了一种称为经验小波变换(EWT)的自适应信号处理方法.其核心思想是通过对信号的Fourier谱进行自适应划分,建立合适的小波滤波器组来提取信号不同的AM-FM成分.-Empirical wavelet transform
提取语音信号基频
- 用自相关函数提取语音信号基频，提取音频文件的基频等高线(Use the autocorrelation function on segments of the signal (windowsize: 100ms) and compute the fundamental frequency. Use a max_time_lag of 100ms in the autocorrelation function and a window shift of 25ms. Create a fundame
matlabyuyin
- matlab在语音信号处理中的使用，包括基频提取，语音识别，语音增强等(The application of MATLAB in speech signal processing including pitch extraction, speech recognition, speech enhancement and so on.It uses matlab.)
short_zero_crossing_rate
- 语音信号的短时过零率求取，此过程为语音信号的一个重要特征提取过程，可用于端点检测等过程。(This process is an important feature extraction process for speech signals and can be used for endpoint detection and other processes.)
matlab
- 这是一个具有语音的采集、读取、内插恢复、重采样，语音的时域参数的计算、端点的检测、基音周期的提取，语音的加噪、滤波及每次处理后语音的播放等功能的语音信号处理系统。(speech signal processing)
LPCC
- 线性预测倒谱系数（Linear Prediction Cepstrum Coefficient,LPCC）是线性预测系数（Linear Prediction Coefficient,LPC）在倒谱域中的表示。该特征是基于语音信号为自回归信号的值设，利用线性预测分析获得倒谱系数。(Linear Prediction Cepstrum Coefficient)
yuchuli1
- 基于python平台的语音信号的预处理和MFCC39维度的特征提取(MFCC based on python)