▍1. g729codec
g729编码解码器,挺好的用于学习音频编解码入门(g729 codec)
语音信号的矢量量化,其中vector_quantization.m为读取音频文件,转换成语音帧,求lpc及lsf参数以及进行矢量量化的主程序,a_lsf_conversion.m为主程序调用的lpc转lsf参数的子程序,INLS1.wav为音频文件。(Vector quantization of speech signals, wherein vector_quantization.m to read audio file, converted into speech frames, seeking lpc and lsf parameters and vector quantization of the main program, a_lsf_conversion.m main program calls a subroutine parameter lsf lpc turn, INLS1.wav as an audio file.)
富迪FM2018语音净燥及消侧音的专用IC开发资料,pdf格式,找了好久才找到的东西,(Fudi FM2018 voice net dry and antisidetone dedicated IC development information, pdf, looking for a long time to find something,)
语音压缩 自适应差分脉冲调制编码 即ADPCM 用matlab开发 (Adaptive differential pulse-modulated voice compression coding using matlab development that ADPCM)
滤波器设计 产生一个连续信号,包含低频,中频,高频分量,对其进行采样,进行频谱分析,分别设计低通,带通,高通滤波器对信号进行滤波处理,观察滤波前后信号的频谱。(The filter is designed to generate a continuous signal containing the low, mid, and high frequency components, to be sampled, for spectral analysis, respectively, to design a low-pass, band-pass filter processing is performed by a high-pass filter on the signal, the spectrum of the signal of the observation before and after filtering.)
关于迭代次数对turbo码性能的影响。很适用,希望大家下载。(About the number of iterations of the turbo code performance. Very applicable, I hope you download.)
迁移学习程序,适用于NLP问题,大家可以看看。(To migrate learning program suitable for NLP problems)
G.711是一种由国际电信联盟(ITU-T)订定音频编码方式,又称为ITU-T G.711。 主要用于音频的编解码。(G.711 )
A率13折线的编译码实现 1、了解PCM及13折线A率编码及译码的原理; 2、随机给出一个语音信号,并用A率13折线PCM对其编码; 3、将编码后的信号经过加性高斯白噪声信道,并在接收端对其进行译码; 4、分别画出原始语音信号、PCM编码信号以及译码后信号的波形。 (A rate of 13 polyline encoding and decoding implementation 1, to understand the principles of PCM and the broken line 13 A rate of encoding and decoding Two were given a voice signal, and PCM encode A rate of 13 polylines 3, the encoded signal through an additive white Gaussian noise channel, and the receiver decodes 4, respectively, to draw the waveform of the original speech signal, the PCM encoded signal and decoding signals.)
Speech Technology: A Practical Introduction Topic: Spectrogram, Cepstrum and Mel-Frequency Analysis Kishore Prahallad Email: skishore@cs.cmu.edu Carnegie Mellon University & International Institute of Information Technology Hyderabad
MMSE语音降噪算法,使用维纳滤波器的方法,运算量适中,能实时处理(MMSE weiner noise reduction)
quatization of audio signal and plot the signal after quantization
This is file to compress Audio using CVSD at 16 Kbps
信号的hilbert变换,C++代码,不用matlab的可试试(the c++ code for the hilbert transform)
说明: 《VC视频音频编解码技术及实践》书的光盘代码(" VC video audio codec technology and the practice of" book on CD-ROM code)
dtmf 拨号程序,Pocket PC ,vc 工程演示(dialDTMF a Pocket PC (Pocket Contact) plugin to dial telephone numbers using DTMF s (aka Dual Tone Multi-Frequency) (tested under PPC2000 MIPS, PPC2000 ARM, PPC2002 ARM, Xscale WMPPC2003 and WM2003SE, WM5- if you want your processor supported either build the exe s yourself from CVS or email me and I may look into it). There is a listed of known working hardware in the FAQ.)