GMM

mfcc.m 用于语音识别上的mfcc特征参数提取函数。(mfcc.m mfcc characteristic parameters for speech recognition on the extraction function.)

基于matlab的HMM语音识别仿真，通过matlab实现对0-9数字的语音识别(HMM speech recognition simulation based on MATLAB)

基于lms的改进型谱减法，降低噪声效果好(Improved LMS-based spectral subtraction, noise reduction effect)

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.)

H-H微分方程的matlab原代码,内包含说明文档,对模拟神经元有帮助.(H-H Differential Equations Matlab original code, which includes documentation, simulated neurons to help.)

语音信号的特征提取，语音信号的Mel倒谱特征（MFCC）的求解方法，语音信号的线性预测原理以及LPC特征的求解方法 (Speech signal feature extraction, speech signal characteristics of Mel cepstrum (MFCC) the solution, voice signals, as well as the principle of linear prediction characteristics of the LPC method)

基于最小均方误差的助听器语音去噪算法, 用于以最大程度消除助听器接受的语音杂音.(Speech denoising algorithm based on the minimum mean square error of a hearing aid for the hearing aid acceptable voice murmur eliminated to the greatest extent.)

语音合成程序源码！psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换，的到短时谱与短时谱包络。短时谱除以短时谱包络的到声源短时谱，对声源短时谱的实部与与虚部分别进行线性插值，就能达到改变变语音信号基频的目的，然后再进行频域反变换，可的到变换后的短时语音信号。短时谱包络部分也能独立改变，以达到改变音色的目的。 (Voice synthesis program source! psalo frequency domain pitch synchronous superposition method. It was first carried out on the original speech signal a short-time frequency-domain transform, to the short-time spectrum and short-time spectrum envelope. The short-time spectrum divided by the short-time spectrum envelope of the short time spectrum of the sound source, the short time spectrum of the real part of the sound source, and the imaginary parts of the linear interpolation, can achieve the purpose of changing the fundamental frequency of the alternating speech signal, and then then the inverse transform of the frequency domain, can be to the short-time speech signal after conversion. The short-time spectral envelope section can be varied independently, in order to achieve the purpose of changing the tone.)

iLBC Speech Coder ANSI-C Source Code WebRtcIlbcfix_CbSearch.h.

基于谱减法的语音降噪处理，里面有声音文件，可以直接运行观察效果(Based on the spectral subtraction speech denoising processing, there are sound files, can be run directly observed effect)