Welcome![Sign In][Sign Up]
Location:
Search - voice quality

Search list

[Voice CompressG.729_Voice

Description: 感觉语音质量还可以!大家看看!-Voice quality can also feel! Let us take a look!
Platform: | Size: 133120 | Author: 孔猛 | Hits:

[ICQ-IM-ChatNetTalk

Description: 聊天程序,由语音和视频的功能。无论界面还是程序质量都非常高. -Chat program, from voice and video functions. Regardless of interface or process quality is very high.
Platform: | Size: 4605952 | Author: Joke | Hits:

[Audio programPESQ

Description: PESQ,用于语音质量评价的客观标准,本论坛已有的是MATLAB源码,现在上载的是标准C代码!-PESQ, for an objective evaluation of voice quality standards, this forum is the MATLAB source code has now uploaded the standard C code!
Platform: | Size: 38912 | Author: junyong | Hits:

[Windows DevelopT-REC-P.862-200102-I!!SOFT-ZST-E

Description: ITU-T P.862 即PESQ语音质量评价的源程序,内有PESQ的说明文档-ITU-T P.862 that PESQ voice quality evaluation of source code, with documentation of the PESQ
Platform: | Size: 5801984 | Author: 刘文旭 | Hits:

[Compress-Decompress algrithmsilbc

Description: Ilibc 语音编解码库算法。语音质量高。接口简单。-Ilibc voice codec algorithm library. High voice quality. Simple interface.
Platform: | Size: 370688 | Author: 林群阳 | Hits:

[ActiveX/DCOM/ATL119128990673-760

Description: This ETSI TETRA codec.It enable Intercommunication between TETRA and other 3G networks without transcoding, and to provide enhanced voice quality for TETRA by using the latest low bit rate voice codec technology.
Platform: | Size: 889856 | Author: 黄桃园 | Hits:

[Multimedia DevelopPESQ

Description: PESQ源代码,实现PESQ语音质量分析功能,VC下编译通过-PESQ source code, to achieve PESQ voice quality analysis function, VC under the compiler through
Platform: | Size: 43008 | Author: GaoXin | Hits:

[BooksPESQandothers

Description: 语音质量客观评价的一些论文资料,以pesq方面的为主-Objective evaluation of voice quality papers information to the main aspects of PESQ
Platform: | Size: 2198528 | Author: 陶凯 | Hits:

[Multimedia programCVSD

Description: CVSD编码的语音质量受跳频速率影响研究 是研究语音编码和提高语音质量的好-CVSD coded voice quality affected by the frequency hopping rate speech coding research is to study and improve voice quality good Dongdong
Platform: | Size: 163840 | Author: 杨帆 | Hits:

[Windows Developwav_analyze

Description: 正弦分析程序,用于通信语音质量分析,C语言编程,源代码-Sinusoidal analysis procedures for communication of voice quality analysis, C language programming, source code
Platform: | Size: 2380800 | Author: GaoXin | Hits:

[Internet-NetworkSkype

Description: Skype是创建Kazaa的组织在2003年开发的一个基于Peer-to-Peer(对等网络)的VoIP客户端。它可以几乎无缝的穿越NAT和防火墙,并且语音质量比其他的VoIP客户端软件要好很多。他加密了端到端的通话,分散式存储用户信息,支持即时消息通信和网络语音会议。 -Skype are creating Kazaa organization developed in 2003 based on Peer-to-Peer (peer-to-peer networks) VoIP client. It can be almost seamless across NAT and firewall, and voice quality than other VoIP client software much better. He had end-to-end call encryption, decentralized storage of user information in support of instant messaging and online voice communication session.
Platform: | Size: 221184 | Author: wj | Hits:

[Internet-Networkgsm

Description: 这个网络电话程序是linux下,用C语言实现的。它既不是实现的H.323 或 SIP协议, 也没有使用RTP协议,更没有使用到任何其它第三方软件,不过,它确实工作的很好。通话话音质量非常不错。-The network telephone program is under linux, using the C language. It is not the realization of the H.323 or SIP protocol, but also did not use the RTP protocol, but did not use to any other third-party software, but it does work well. Very good voice call quality.
Platform: | Size: 59392 | Author: iy.i | Hits:

[Internet-Network514613687

Description: 使用G.729协议压缩的语音传输程序 感觉语音质量还可以!大家看看 -The use of G.729 compressed voice transmission protocol procedures for voice quality can be felt! Let us take a look
Platform: | Size: 129024 | Author: rweew | 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:

[Multimedia Developv_chat

Description: 一款性能良好的视频会议系统;可以实现视频,语音,文本信息的交流。需要外部设备:摄像头,耳机+麦克。经测试,图像清晰,语音质量可靠,图像音频传输失真很小,速度可靠。是多媒体网络传输开发者良好的参考。 -A good performance video conferencing system can be video, voice, text information. The need for external equipment: camera, headset+ Mike. Tested and the pictures are clear, reliable voice quality, image audio transmission distortion is very small, reliable speed. Multimedia network transmission developers a good reference.
Platform: | Size: 3747840 | Author: 路人 | Hits:

[Internet-NetworkVS2005sipcode

Description: SIP它既不是会话描述协议,也不提供会议控制功能。为了描述消息内容的负载情况和特点,SIP 使用 Internet 的会话描述协议 (SDP) 来描述终端设备的特点。SIP 自身也不提供服务质量 (QoS),它与负责语音质量的资源保留设置协议 (RSVP) 互操作。它还与若干个其他协议进行协作,包括负责定位的轻型目录访问协议 (LDAP)、负责身份验证的远程身份验证拨入用户服务 (RADIUS) 以及负责实时传输的 RTP 等多个协议。-It is not a SIP Session Description Protocol, nor the provision of conference control functions. In order to describe the content of the message and the characteristics of the load, SIP to use Internet-Session Description Protocol (SDP) to describe the characteristics of terminal equipment. SIP itself does not provide Quality of Service (QoS), it is responsible for voice quality and resource reservation setup protocol (RSVP) interoperability. It also with a number of other collaboration agreements, including positioning of the Lightweight Directory Access Protocol (LDAP), is responsible for authentication of the Remote Authentication Dial-In User Service (RADIUS) as well as real-time transmission of multiple protocols such as RTP.
Platform: | Size: 369664 | Author: 离开 | Hits:

[OtherQualityofServiceProvisioning

Description: —One of themajor challenges in supportingmultimedia services over Internet protocol (IP)-based code-division mul- tiple-access (CDMA) wireless networks is the quality-of-service (QoS) provisioning with effi cient resource utilization. Compared with the circuit-switched voice service in the second-generation CDMA systems (i.e., IS-95), heterogeneous multimedia applica- tions in future IP-based CDMA networks require more complex QoS provisioning and more sophisticated management of the scarce radio resources. This paper provides an overview of the CDMA-related QoS provisioning techniques in the avenues of packet scheduling, power allocation, and network coordination, summarizes state-of-the-art research results, and identifi es further research issues.
Platform: | Size: 408576 | Author: Duc Long | Hits:

[TCP/IP stackPhone

Description: 网络多媒体通信 1、编制一个网络多媒通信软件,实现: 在发送端采集话筒声音,通过网络实时传输到接收端,并在接收端播放出来。 2、通过使用TCP、UDP、变更分组大小来对比收发端声音同步情况及播放质量。 本实验技术不同于课上所讲的回调函数,利用了MFC的消息处理机制,用消息处理函数替代了回调函数,但整个流程是一样的。本程序采用C/S模式,其中Server端为项目PhoneToFile,Client端为项目Client,Server端的功能为采集声音数据并发送给客户端,Client端将收到的声音数据播放。在测试中只需在Server端打开Server程序并播放音乐或用话筒录音,在Cliet端打开Client程序,用耳机就可以听到音乐或录音。-Internet Multimedia Communications 1, the preparation of a network of multi-media communications software to achieve: In the transmitter microphone capture sound, real-time transmission through the network to the receiving side, and playing out at the receiving end. 2, using TCP, UDP, change the packet size to compare the situation and simultaneously send and receive-side audio playback quality. The experimental technique is different from the class talked about the callback function, use of MFC s message handling mechanism is replaced by the message handler callback function, but the whole process is the same. This program uses C/S mode, in which Server-side for the project PhoneToFile, Client-side for the project Client, Server-side functionality for the capture audio data and sent to the client, Client-side will receive the voice data playback. In the test, simply open the Server in the Server-side program and play music or microphone recording, open the Client program
Platform: | Size: 66560 | Author: zym | Hits:

[Windows DevelopNGNNETWORKTEST

Description: 探讨了NGN网络测试中的性能测试技术,重点阐述了呼叫性能测试技术;介绍了语音质量的测试原理、测试规范、测试方法和测试参数;对NGN网络的维护进行了比较简单的讨论,阐述了运营商的两种网络维护模式的相同和不同之处,以及如何应用。-Discusses the NGN network testing the performance test technology, focusing on the call performance test techniques introduced the voice quality of the test theory, test specifications, test methods and test parameters for the maintenance of NGN networks were compared to a simple discussion, elaborated The two network operators in maintenance mode the similarities and differences, and how to apply.
Platform: | Size: 8192 | Author: 张容 | Hits:

[OtherGSM

Description: 《网规网优案例集锦V1.0》是近两年来工程师网规网优经验的总结,有些甚至是现场工作中的教训。通过学习本案例集锦,有助于提高工程师现场处理问题的能力,增加网规网优知识和经验,也可以使工程师避免犯同样的错误! 《网规网优案例集锦V1.0》将全文共168篇案例划分成15个专题:仪器使用、数据配置和动态设定、覆盖、拥塞、话务均衡、干扰、掉话、切换、手机上网和呼叫、寻呼和指配、话音质量、信号波动、用户不在服务区、GPRS、其它,方便工程师查阅和借鉴! 无线网络规划技术支持组 2002年6月 -"Net Regulation Network Optimization Case Collection V1.0" the past two years, network engineers, network optimization and experience in regulation, and some even on-site work in lessons. Through the study of this case highlights to help improve the engineer on-site problem solving skills, increase net regulatory network optimization knowledge and experience, also allows engineers to avoid the same mistakes! "Net Regulation Network Optimization Case Collection V1.0" A total of 168 cases of full-text is divided into 15 topics: instrumentation, data configuration and dynamic settings, coverage, congestion, traffic balance, interference, dropped calls, switching, mobile Internet access and call, paging, and assignment, voice quality, signal fluctuation, the user is not in service area, GPRS, Other, to facilitate engineers to access and learn from!
Platform: | Size: 1628160 | Author: lenald | Hits:
« 1 2 34 5 6 7 8 9 10 ... 19 »

CodeBus www.codebus.net