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newpnn
- 基于GMM的概率神经网络PNN具有良好的泛化能力，快速的学习能力，易于在线更新,并具有统计学的贝叶斯估计理论基础，已成为一种解决像说话人识别、文字识别、医疗图像识别、卫星云图识别等许多实际困难分类问题的很有效的工具。而且PNN不但具有GMM的大部分优点，还具有许多GMM没有的优点，如强鲁棒性，需要更少的训练语料，可以和其他网络其他理论无缝整合等。-GMM based probabilistic neural network PNN good generalization ability, the
wavesurfer-185-win
- 一个与htk配合使用的声音处理的软件,剑桥的,很不错,可以将wav等格式的文件转化为htk格式的文件-a tie with the use of voice processing software, Cambridge, very good, as can be wav files into a document format HTK
WavPlayer
- to show the waveform of audio file and play it on computer Purpose: Familiar with WAV file format and UI design It should have the following functions: Provide a Graphic User Interface for user to browse the file system and select one WAV file
audiofile-0_2_6
- The Audio File Library provides a uniform programming interface to standard digital audio file formats. This library allows the processing of audio data to and from audio files of many common formats (currently AIFF, AIFF-C, WAVE, NeXT/Sun
rtp2pcm
- 將RTP轉成PCM/MP3,方便大家測試RTP封包內容是否如預期-RTP will turn into PCM/MP3 facilitate testing RTP packet contents as expected
yinpintongxin
- 本程序是一个基于Visual C++的很好的语音通信实例,只要稍微修改即可作为工程应用,有很高的使用价值-this is a process based on Visual C is a good example of the Speech Communication, As long as can be modified slightly as the project application, a high value
g729A
- 本程序的压缩编码是G.729的编解码程序，其的算法是基于CS-ACELP的算法基础上的编码。用在语音通信中，例如手机中-This program s arithmetic is base on ITU G.729 Speech Vocoder, which is a CS-ACELP compressing coder, usually use in speech comunication, such as mobile phone.
G723[1].1
- 本程序的压缩编码是G.723的编解码程序，其的算法是基于ACELP的算法基础上的编码。用在语音通信中，例如手机中-the procedure coding is the G.723 codec procedure, the algorithm is based on ACELP algorithm based on the coding. Used in voice communications, such as mobile phones
QAM_Matlab 软件无线电相关程序
- 软件无线电相关程序，如信号产生，抽样，压缩，编码，自动争议控制-Software radio procedures, such as signal generation, sampling, compression, coding, automatic control controversy
G711G721G723.rar
- 用C语言编程实现的G.711.721.723标准,可以作为调试用,值得作为参考下载,Implementation using C language programming of G.711.721.723 standards, can be used as a debugger, it is worth as a reference download
G711Codec.zip
- Delphi版的G711语音编解码模块，做VOIP之类的应用可以用到！,Delphi version of the G711 voice codec module, so applications such as VOIP can be used!
dtmf
- 本程序用于检测音频文件中是否具有DTMF信号，若有则将其检出。 程序首先使用Goertzel算法求出以FRAMESIZE（默认200）为大小的一帧数据在8个DTMF频点上的能量。 对Goertzel算法的改进，对于系数的计算不是采用2*cos[2*pi*k/N]，而是采用2*cos[2*pi*fn/fs],这样能够降低误差。 确定了8个频点的能量后运用一系列判决门限来确定有没有DTMF信号，以及信号是什么。 -This procedure used to detect wh
lab12-Audio
- 利用A律十三折线算法进行压缩和解压缩，实现音频信号的压缩和解压缩-make advantage of the A law to despread and spread the signal, so as to despread and spread the audio signal
intercom-latest.tar
- Intercom 是一个 Unix系统上灵活的语音传输软件。支持标准音频压缩比如GSM, G.711, and G.72x和其他音频编码。Intercom专为高速网络设计来传输高品质的语音，也支持窄带传输-Intercom is a Unix system flexible voice transmission software. Support audio compression standards such as GSM, G.711, and G.72x and other audio c
103244815yuyinyasuo
- 语音压缩，A律语音压缩以及C语言下的编程方法-Voice compression, A Law of voice compression, as well as C programming language under the Ways
mmamrdmx_0.9.0.1_src
- MONOGRAM AMR Splitter v0.9.0.1-I’ve noticed on the Codec Guide discussion board that some people were missing an AMR parser filter so I’ve decided to write one. It follows the same concept as the musepack splitter filter so it was basically just copy
mmamre_1.0.0.0_src
- MONOGRAM AMR Encoder v1-As the post title says my today’s filter is a free implementation of AMR DirectShow filter. It is based on the source code provided by 3GPP.org site- the same as libamr_nb and comes as GNU GPL.
audio6701
- 基于TI 6711的 语音信号处理程序，包括应用软件中断的语音信号处理实例，以及音频流编、解码算法。-TI 6711 based on the speech signal processing procedures, including application software interrupt voice signal processing examples, as well as for the audio stream, decoding algorithm.
Cpp1
- 本来想做一个BPSK的调制和解调的混合编程实验，由于混合编程的库里面没有butter、cheby1等滤波器函数，用C编又不会，所以只好请教高手了。呵呵~-Originally wanted to do a BPSK modulation and demodulation of the mixed-programming experiment, as a result of mixed programming library There are no butter, cheby1 function
MELP4
- This standard describes the interoperability requirements relating to the conversion of analog voice to 2,400 bits/s digitized voice by a method known as Mixed Excitation Linear Prediction (MELP) and reconversion back to analog voice. An algorith