ITU-T G.729語音壓縮算法。 description: Fixed-point description of commendation G.729 with ANNEX B Coding of SPEECH at 8 kbit/s using Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP) with Voice Activity Decision(VAD), Discontinuous Transmission(DTX), and Comfort Noise Generation(CNG).
標簽： description commendation Fixed-point 729
上傳時間： 2014-11-23
上傳用戶：thesk123
Hidden_Markov_model_for_automatic_SPEECH_recognition This code implements in C++ a basic left-right hidden Markov model and corresponding Baum-Welch (ML) training algorithm. It is meant as an example of the HMM algorithms described by L.Rabiner (1) and others. Serious students are directed to the sources listed below for a theoretical description of the algorithm. KF Lee (2) offers an especially good tutorial of how to build a SPEECH recognition system using hidden Markov models.
標簽： Hidden_Markov_model_for_automatic SPEECH_recognition implements left-right
上傳時間： 2016-01-23
上傳用戶：569342831
G729編碼和解碼的Matlab源程序 調用方法： 1.需要編碼時，在 Matlab命令行下輸入"G729code(SPEECH)"(請務必保證 SPEECH是已經存在在 Matlab工作區的一個行向量，其中是語音數據)，然后你會在你的文件夾下看到一個"coding.bit"的文件。 2.需要解碼時，把"coding.bit"的文件復制到"G729 decoder"的目錄下，在Matlab命令行下輸入"G729decodermain"，然后你會在文件夾下看到"syn_SPEECH.wav"，那就是重構語音。 enjoy it!
上傳時間： 2013-12-22
上傳用戶：GavinNeko
Generalized Mel frequency cepstral coefficients for large-vocabulary Speaker-Independent Continuous-SPEECH Recognition 關于MFCC算法的很好的英語文章
標簽： Speaker-Independent large-vocabulary coefficients Generalized
上傳時間： 2014-01-23
上傳用戶：liglechongchong
AutoSummary uses Natural Language Processing to generate a contextually-relevant synopsis of plain text. It uses statistical and rule-based methods for part-of-SPEECH tagging, word sense disambiguation, sentence deconstruction and semantic analysis.
標簽： contextually-relevant AutoSummary Processing Language
上傳時間： 2014-01-08
上傳用戶：zhangzhenyu
DM6446的codec engine 應用端算法源碼，包括video，audio，SPEECH等開發實例。
上傳時間： 2014-01-17
上傳用戶：dbs012280
Digital Signal and Image Processing Using MATLAB The most important theoretical aspects of image and signal processing (ISP) for both deterministic and random signals are covered in this guide to using MATLAB® . The discussion is also supported by exercises and computer simulations relating to real applications such as SPEECH processing and fetal-heart–rhythm tracking, and more than 200 programs and functions for numerical experiments are provided with commentary.
標簽： theoretical Processing important Digital
上傳時間： 2016-07-08
上傳用戶：z1191176801
Matsig is an object-oriented signal class library for MATLAB 6.5 and later. It implements a signal class, simplifying operations and manipulations common in audio signal processing and SPEECH processing
標簽： signal object-oriented implements library
上傳時間： 2014-12-21
上傳用戶：lunshaomo
With the advent of multimedia, digital signal processing (DSP) of sound has emerged from the shadow of bandwidth-limited SPEECH processing. Today, the main appli- cations of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including percep- tual measurement techniques and analysis/synthesis methods. Smaller but nonetheless very important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application oriented research.
標簽： multimedia processing the digital
上傳時間： 2014-01-23
上傳用戶：xwd2010
Many problems in statistical pattern recognition begin with the preprocessing of multidimensional signals, such as images of faces or spectrograms of SPEECH.
標簽： multidimensional preprocessing statistical recognition
上傳時間： 2016-12-01
上傳用戶：515414293
