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[Disk ToolsUTalk2.0

Description: UTalk是一款主要针对网络游戏用户和局域网游戏用户而开发的团队语音通信工具。-UTalk is mainly aimed at network game users and LAN game users and the development team Speech Communication tools.
Platform: | Size: 4368343 | Author: 无心 | Hits:

[Voice Compressyinpintongxin

Description: 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
Platform: | Size: 149846 | Author: qiusha | Hits:

[Multimedia DevelopVisual C++视频音频开发实用工程案例精选

Description: 这是《Visual C++视频/音频开发实用工程案例精选》一书的源代码。精选了大量的具有商用价值的工程案例,包括视频捕捉系统、视频会议系统和远程视频监控系统的开发技术;网络、多媒体技术的底层实现;MP3深入编程技术等。最后一章为基于IBM ViaVoice的语音识别系统在上位机和下位机之间的通信的架构及实现。-This is the "Visual C video/audio develop practical projects selected cases," a book source. Featured in a great deal of value to the business case works, including video capture systems, video conferencing systems and remote video surveillance systems development; Network, multi-media technology to achieve the bottom; MP3-depth programming skills. The final chapter of the IBM ViaVoice speech recognition system in the PC and the next crew of communication between the organization and realization.
Platform: | Size: 22413312 | Author: 靳晓辉 | Hits:

[Disk ToolsUTalk2.0

Description: UTalk是一款主要针对网络游戏用户和局域网游戏用户而开发的团队语音通信工具。-UTalk is mainly aimed at network game users and LAN game users and the development team Speech Communication tools.
Platform: | Size: 4368384 | Author: | Hits:

[Voice Compressyinpintongxin

Description: 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
Platform: | Size: 190464 | Author: qiusha | Hits:

[Communication-MobileSpeechcodingtechnologyanalysisandapplication

Description: 对语音编码技术的特点进行了分析与研究, 对波形编码、声码器和混合编码三种主要 的语音编码进行了比较, 并介绍了GSM 和CDMA 两种系统语音编码及部分常用语音编码芯 片, 指出了语音编码技术的发展趋势, 并根据矿井井下特点, 对矿井通信系统所采用的语音 编码技术进行了研究, 最后得出了几点有用的结论.-Of speech coding technology characteristics of the analysis and research on waveform coding, vocoder and mixed encoding three major speech coding are compared, and introduced the GSM and CDMA both types of systems commonly used part of speech coding and speech coding chip pointed out that the speech coding technology, development trends, and in accordance with the characteristics of the mine pit of the mine communication system used in speech coding technology for the study came to the conclusion that some useful conclusions.
Platform: | Size: 188416 | Author: 王萍 | Hits:

[Speech/Voice recognition/combineDigital_Speech

Description: Digital Speech:Coding for Low Bit Rate Communication Systems Second Edition 经典语音编码书籍-Digital Speech: Coding for Low Bit Rate Communication Systems Second Edition classic speech coding books
Platform: | Size: 8531968 | Author: 计算机 | Hits:

[Speech/Voice recognition/combineScrap

Description: The speech signal contains rich messages, and three main recognition fields from speech signal, which are of most interest and have been studied for several decades, are speech recognition, language recognition and speaker recognition. In this the focus is on speech recognition field. In our everyday lives there are many forms of communication, for instance: body language, textual language, pictorial language and speech, etc. However amongst those forms speech is always regarded as the most powerful form because of its rich dimensions character. Except for the speech text (words), the rich dimensions also refer as the gender, attitude, emotion, health situation and identity of a speaker. Such information is very important for an effective communication. From the signal processing point of view, speech can be characterized in terms of the signal carrying message information. The waveform could be one of the representations of speech, and this kind of signal has been most useful in practical applications.
Platform: | Size: 3072 | Author: kinny garg | Hits:

[OtherPattern_Recognition_Theodoridis_Koutroumbas

Description: The philosophy of the book is to present various pattern recognition tasks in a unified way, including image analysis, speech processing, and communication applications. Despite their differences, these areas do share common features and their study can only benefit from a unified approach.-The philosophy of the book is to present various pattern recognition tasks in a unified way, including image analysis, speech processing, and communication applications. Despite their differences, these areas do share common features and their study can only benefit from a unified approach.
Platform: | Size: 10775552 | Author: Andrey | Hits:

[Speech/Voice recognition/combinewavelet

Description: SPEECH ENHANCEMENT BASED ON WAVELET DENOISING Abstract: - Noise is an unwanted and inevitable interference in any form of communication. It is non-informative and plays the role of sucking the intelligence of the original signal. Any kind of processing of the signal contributes to the noise addition. A signal traveling through the channel also gathers lots of noise. It degrades the quality of the information signal. The effect of noise could be reduced only at the cost of the bandwidth of the channel which is again undesired as bandwidth is a precious resource. Hence to regenerate original signal, it is tried to reduce the power of the noise signal or in the other way, raise the power level of the informative signal, at the receiver end this leads to improvement in the signal to noise ratio (SNR). There are several ways in doing it and here the focus is on adaptive Signal processing new technique (Grazing Estimation method) to improving the signal to noise ratio.-SPEECH ENHANCEMENT BASED ON WAVELET DENOISING Abstract:- Noise is an unwanted and inevitable interference in any form of communication. It is non-informative and plays the role of sucking the intelligence of the original signal. Any kind of processing of the signal contributes to the noise addition. A signal traveling through the channel also gathers lots of noise. It degrades the quality of the information signal. The effect of noise could be reduced only at the cost of the bandwidth of the channel which is again undesired as bandwidth is a precious resource. Hence to regenerate original signal, it is tried to reduce the power of the noise signal or in the other way, raise the power level of the informative signal, at the receiver end this leads to improvement in the signal to noise ratio (SNR). There are several ways in doing it and here the focus is on adaptive Signal processing new technique (Grazing Estimation method) to improving the signal to noise ratio.
Platform: | Size: 192512 | Author: majid | Hits:

[Multimedia DevelopAn_Adaptive_Jitter_Buffering_Algorithm_for_Voice_o

Description: 当IP语音包的网络时延抖动较小时,一般的语音缓冲算法可以得到较好的语音质量。当网络中存在突发大时延时,就会出现极大丢包率或极大端到端时延,从而难以获得好的语音质量。为此,提出针对突发大时延下的自适应语音缓冲算法。通过估算网络平均时延和学习语音包经过的网络路径上的状态,来确定需要控制端到端时延大小和语音包的丢包率,动态调整Jitter Buffer队列的最小深度和最大深度,从而可以尽量减小语音裂缝(gap)的出现。通过基于听觉模型的客观音质评价(PESQ)仿真计算以及在实际语音网关设备中的应用表明算法对语音通信质量有一定的改善作用。-The continuous playout of voice packets in the presence of variable network delays is often achieved by buffering the received voice packets for sufficient time. Basic jitter buffering algorithms can work well only when the delay does not spike status of the networks, is presented to promote the quality of voice communication. It timely adjusts the minimal and maximal depth of buffer queue according to the control target of end-to-end delay and packet loss rate. The algorithm can much more easily achieve the continuous playout because it plays voice packet at a fixed inter-play time in the most time of a talk-spurt. The control target of packet loss rate can be extended to 20 . However, the basic algorithms can only bear 5-10 of the packet loss rate. Perceptual evaluation of speech quality(PESQ) is applied to assess the speech quality in the simulation. It is shown that the algorithm can obviously promote the quality of voice communication in IP networks with spike delay. The practic
Platform: | Size: 329728 | Author: 瞿志超 | Hits:

[Delphi VCLnrCommLib

Description: nrComm Lib Pro 8.29 for Delphi & C++ Builder 7-2010 nrComm Lib provides some tools for performing the serial communications tasks in Delphi/CBuilder development. It has ready solutions for many communication tasks : RS232, TAPI (data and voice modems), Text to speech conversion, access to LPT port, barcode readers, Bluetooth connections and devices, work with GSM (sms send receive, access to phonebook etc.), ZModem file transfer protocol, HID devices, Terminal, logging and other more.
Platform: | Size: 25238528 | Author: Sjaak Willem | Hits:

[OtherMicrophone_Array_Signal_Processing

Description: 语音阵列信号处理,对于sonar检测和水声通信是很好的参考-Speech array signal processing for sonar detection and underwater acoustic communication is a good reference
Platform: | Size: 2705408 | Author: loren.he | Hits:

[OtherDPCM

Description: 通信原理课程设计:语音信号基带通信传输系统仿真基于DPCM编码和循环码 -Communication Theory Course Design: speech signal base-band telecommunication transmission system simulation Based on DPCM coding and cyclic codes
Platform: | Size: 512000 | Author: 黄蓉 | Hits:

[Program docSpeech-Recognition-HOWTO

Description: pdf compressed format of communication speech
Platform: | Size: 31744 | Author: ch ravinder | Hits:

[SCMData-Transmission-over-Speech-Coded-Voice-Channels

Description: The voice channel in mobile communication systems have high priority and are almost always available. By using the voice channel also for data transmissions it is possible to get the same availability as for voice calls. But due to speech codecs in the voice channel, regular modems can not be used and special techniques are needed to transmit data. This thesis presents methods to transmit data over the voice channel in a GSM, UMTS or TETRA network. The focus has been on robust data transmission rather than high data bit rates. Approaches are introduced which improve the reliability for transmissions even for systems with low rate speech codecs and channels with some distortion. The results of the thesis are suggestions of symbol patterns and ways to create and adapt symbols for specific application and channel conditions to achieve the desired goal for the application
Platform: | Size: 688128 | Author: 阿铁 | Hits:

[Speech/Voice recognition/combineSpeech-Digital-Signal-Processing

Description: 本书就是说明怎样才能把数字信号处理的技术应用于语音通信有关的问题。-This book is to explain how the digital signal processing technology for voice communication-related issues.
Platform: | Size: 9345024 | Author: dustfly | Hits:

[matlabprogress-in-speech-compression

Description: 为了满足数字通信及其它商业应用的需求,语音压缩编码技术得到了迅速发展。介绍了目前语音压缩编码技术 的研究进展,主要包括连续可变斜率增量调制(CVSD)、小波分析、多脉冲激励线性预测编码、散布脉冲码激励线性预 测(DP-CELP)、多重脉冲散布非均匀代数码本激励线性预测(MPD-USACELP)、波形内插(Ⅷ)、线谱对(频率)(LSP)的量化-In order to satisfy deman凼of the digital communication and other commercial applicatiOils, the speech compression technology has been developed rapidly.The present research progress in speech compression technology is introduced in this paper including CVSD, wavelet analysis and its application to speech coding,MPLPC,DP-CELP,MPD-USACELP,Ⅷand quantification of LSF.These algorithins
Platform: | Size: 309248 | Author: 杨亚欣 | Hits:

[Software Engineeringspeech-coummunication-

Description: In this paper we present a robust voice activity detection algorithm which enables further bit rate reduction for practical speech communication systems. -In this paper we present a robust voice activity detection algorithm which enables further bit rate reduction for practical speech communication systems.
Platform: | Size: 199680 | Author: 冯浩 | Hits:

[Industry researchSpeech-Processing.pdf

Description: This document makes an introduction to speech communication, Acoustic Theory of Speech: The Source-Filter Model,Speech Models and Features, Linear Prediction Model of Speech,Harmonic plus Noise Model of Speech, etc.
Platform: | Size: 516096 | Author: prg | Hits:
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