搜索资源列表
iir-c
- 用C语言编写的用双线形变换法实现的IIR滤波器程序,可以参考一下-C language prepared by the double-linear transformation to achieve the IIR filter process can take a look
matlab_yyxhchuli
- mablab语音信号处理,包括波形产生,滤波器分析,滤波器实现,傅立叶变换,Z变换,等语音处理程序-mablab speech signal processing, including waveform generator, filter, filter, Fourier transform, Z transform, and other voice processing
audio_noise
- 语音降噪。从Codec AD50采集话筒语音,通过DSP TMS320vc5402处理,在送到AD50输出降噪后语音,涉及加汉宁窗,切比雪夫滤波器,快速傅立叶变换和反FFT,有声无声判断谱分解,谱合成等功能
MFCC 对输入的语音序列x进行MFCC参数的提取
- 对输入的语音序列x进行MFCC参数的提取,返回MFCC参数和一阶差分MFCC参数,Mel滤波器的阶数为24,fft变换的长度为256,采样频率为8000Hz,对x 256点分为一帧. -The voice of the input MFCC parameters on the sequence of x, return to MFCC parameters and extracted a order difference MFCC parameters, Mel filter for th
matlab_based_adaptive_system_emulation
- 基于matlab的自适应仿真,其算法都是LMS算法,有系统辨识,噪声对消,自适应滤波器,陷波器,图像的DCT变换-adaptive emulation based on LMS,include system recognise,noise canceler,adaptive filter,dct transform
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.
SimulinkFSK
- 题目要求: 1.录制一段自己的语音信号,并对录制的信号进行采样; 2.画出采样后的语音信号的时域波形和频谱图; 3.给定滤波器的性能指标,采用窗函数法和双线性变换法设计滤波器, 并划出滤波器的频域响应; 4.用该滤波器对采集的信号进行滤波,画出滤波后信号的时域波形和频谱, 并对滤波前后的信号进行对比,分析信号的变化; 5.回放语音信号; 6.设计一个信号处理系统界面。-Subject to: 1. Recorded his voice for so
GSM_full_rate.RAR
- 长期预测(LTP)与规则脉冲激励(RPE),而全速率编解码器就被称为RPE-LTP线性预测编码器。 输入至RPE-LTP编码器的数据为包括160个采样值的20ms语音,每一个采样值都拥有13位精度。数据首先通过预加重滤波器来提高信号的高频分量,以获得更好的传输效率。滤波器一般还消除信号上的任何偏移以简化进一步的计算。 正如前面所提到的,语音产生模型可看成是空气通过一组不同大小的圆柱体。短期分析级采用自动相关来计算与模型所用的8个圆柱体有关的8个反射系数,同时采用一种称为S
mfcc_feature_extraction
- 本代码实现语音信号的特征提取功能,包含预加重,加窗,DFT变换,设置滤波器组,计算每隔滤波器输出,求取MFCC系数的全过程-The code feature extraction of speech signal features, including pre-emphasis, windowing, DFT transform, set the filter to calculate every filter output, to strike the entire process of MF
BasedonMATLABspeechsignalspectrumanalysisandfilter
- 录制一段个人自己的语音信号,并对录制的信号进行采样;画出采样后语音信号的时域波形和频谱图;给定滤波器的性能指标,采用窗函数法和双线性变换设计滤波器,并画出滤波器的频率响应;然后用自己设计的滤波器对采集的信号进行滤波,画出滤波后信号的时域波形和频谱,并对滤波前后的信号进行对比,分析信号的变化;回放语音信号-The individual' s own record a voice signal, and the recorded signal is sampled draw sampled
xiaobo
- 语音增强算法及集中经典算法,小波变换法谱减法,滤波器法等-Speech enhancement algorithm and focus on the classical algorithm, wavelet transform spectral subtraction, filter method, etc.
speech-recognition
- 语音识别系统,包括预处理,汉明窗,梅尔频率倒谱率,离散余弦变换,前置滤波器组,对数能量,滤波器组,基本的滤波器组,dtw-speech recognition system,including Pretreatment, Hamming window, Mel frequency cepstrum rate, discrete cosine transform, pre-filter, log energy, filter, basic filter,dtw
yuyingfenxi
- 录制一段自己的语音信号,时间为10s左右,并对录制的信号进行采样;画出采样后语音信号的时域波形和频谱图;采用窗函数法或双线性变换设计滤波器,并画出滤波器的频率响应;然后用自己设计的滤波器对采集的语音信号进行滤波,画出滤波后信号的时域波形和频谱,并对滤波前后的信号进行对比,分析信号的变化;回放语音信号.-Record a voice signal their time was 10s or so, and the recorded signal is sampled draw sampled s
831ccab5cb74
- IIR数字滤波器的设计,通过一定的变换,将模拟滤波器转换成数字滤波器。-IIR digital filter design, an IIR digital filter design, we first write a formula based on indicators of analog filters, and then through some transformation, the conversion formula analog filter into a digital fil
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
dsp
- 本压缩包含解差分方程,z变换,滤波器设计,回响滤波器,采样例子多个dsp的基础知识的相关函数。(Solving difference equation, z transform and filter design)