Search - voice packet

[Audio program] NQOS

Description: VoIP(voice over IP) 就是通过IP 网络承载语音业务,也称IP 网络电话。当网络出现拥塞或传输差错时,语音包就会产生时延、抖动甚至丢失,导致语音不连续或中断,-VoIP (voice over IP) through the IP network to carry voice traffic. also known as IP telephone network. When the network is expected to congestion or transmission errors, the packet will have a voice delay, jitter or loss, voice not lead to continuous or interrupted,
Platform: | Size: 4808 | Author: 张路宜 | Hits:

[Other resource] baoge.netkaiyuabliuyanxitongSQL

Description: 包哥.net开源留言系统 SQL修正版比较好用-Packet Columbia. Net revenue voice mail system earlier version of SQL that handy
Platform: | Size: 148997 | Author: 黄伟文 | Hits:

[Applications] AutoPhoneSys

Description: 自动语音应答系统硬件需要一个语音卡或一个语音modem 其中有6个类，下载包里头有详细类说明-Automatic Voice Response System hardware needs a sound card or a voice modem which six categories, Download Packet Erlitou detailed presentation on
Platform: | Size: 345075 | Author: 陈星 | Hits:

[VOIP program] iad132e

Description: IAD （Integrated Access Device ）132E(T) 综合接入设备（下文简称IAD132E(T) ）是华为技术有限公司下一代网络NGN（Next Generation Network ）解决方案中的重要部件，用以向公司等用户提供小容量VoIP （Voice over IP）/FoIP（Fax over IP ）解决方案。IAD132E(T) 作为VoIP/FoIP 媒体接入网关，应用于NGN 用户接入层，完成模拟话音信号与IP 包之间的转换，并通过包交换网络传送数据的功能，同时还可通过标准MGCP（Media Gateway Control Protocol ）协议，与华为技术有限公司SoftSwitch 软交换设备配合组网，在SoftSwitch 控制下完成主被叫间的话路接续。-IAD (Integrated Access Device) 132E (T) Integrated Access equipment (hereinafter referred IAD132E (T)) is Huawei Technologies Co., Ltd. NGN (Ne Anhui Generation Network) solutions to the important parts, for the company to provide users such as small-capacity VoIP (Voice over IP) / FoIP (Fax o ver IP) solutions. IAD132E (T) as the VoIP / FoIP Media Access Gateway, NGN users for access layer, complete with simulated voice signals between IP packet switching and through packet-switched data transmission network functions, It can also through standard MGCP (Media Gateway Control Protocol ) agreement with Huawei Technologies Co., Ltd. SoftSwitch equipment with soft switch network, SoftSwitch in complete control of the main Called the ground successor.
Platform: | Size: 11576 | Author: 孙昊 | Hits:

[Mathimatics-Numerical algorithms] systerm

Description: 公交车语音系统压缩包里面包括了所有的代码甚至原理图！-Bus packet voice compression system includes all the code even diagram!
Platform: | Size: 439095 | Author: 鲁军波 | Hits:

[Industry research] IP电话的关键技术

Description: ip电话的关键技术,包括语音编码,压缩,打包,分组交换,以及保证语音质量而采取的回声抵消技术.-ip telephone key technologies, including voice coding, compression, packing and packet switching, and voice quality assurance and the echo cancellation technology.
Platform: | Size: 14336 | Author: 周周 | Hits:

[Speech/Voice recognition/combine] VoiceRecogniseCommandMode

Description: 语音识别的例子。请先到微软官方网站下载语音识别包和语音识别SDK，本程序根目录下有一个语法命令文件。-speech recognition examples. Microsoft advised to download the official website of packet voice recognition and voice recognition SDK, The procedures under the root directory is a grammatical order paper.
Platform: | Size: 245760 | Author: kun | Hits:

[Speech/Voice recognition/combine] SpeechMouse

Description: 使用Delphi编写的语音鼠标，请先到微软官方网站下载语音识别包和语音识别SDK。-use Delphi voice mouse, Microsoft advised to download the official website of packet voice recognition and voice recognition SDK.
Platform: | Size: 563200 | Author: kun | Hits:

[VOIP program] rtrytvbi

Description: iLBC 产生背景　　在VoIP的应用中，大部分厂商采用CELP (Code Excited Linear Prediction) 算法的低速率语音编解码，如ITU G.729和G.723.1等。而VoIP应用主要在包交换的IP网络上进行传输，无法避免IP网络的丢包、延时、抖动等实时传输问题，而传统的这几个CELP算法对高丢包的处理不是很好，因而很大程度上会影响语音通话效果。 -iLBC background in the application of VoIP, Most manufacturers to use CELP (Code Excited Linear Prediction) Algorithm low-rate voice codecs, such as G.729 and ITU G.723.1, etc.. And the main application of VoIP in the IP packet-switched network for transmission, could not avoid IP network packet loss, delay, jitter and other real-time transmission, and a few traditional CELP algorithm to handle the high packet loss is not very good, which will largely affect voice calls effect.
Platform: | Size: 67584 | Author: 王刚 | Hits:

[Windows Develop] AudioSample

Description: 一个可以产生G72A和ILBC语音编解码动态库的工程-Can generate a G72A and ILBC voice codec DLL project
Platform: | Size: 873472 | Author: 刘SIR | Hits:

[Delphi VCL] deepblue_Voice_Communicator_Components_v2.5_200704

Description: Voice Communicator (VC) 一套处理音频压缩的的控件包-Voice Communicator (VC) to deal with a set of compressed audio control packet
Platform: | Size: 1611776 | Author: 梁超 | Hits:

[ICQ-IM-Chat] QQSimilar

Description: 这个包不是我开发的，实在淘宝上买的。一个用Delphi7编写的类似于QQ的即时聊天工具，支持文字、语音、视频和分组聊天，功能强大。有客户端和服务端全部源代码，好包含大量的控件且有控件的源代码，如MMTools-This package is not what I developed, it bought on Taobao. Using Delphi7 prepared a similar QQ instant chat tools, support for text, voice, video and packet chat and powerful. There are client and server all the source code, and contains a large number of control and there is control the source code, such as the MMTools
Platform: | Size: 38183936 | Author: 杨淮生 | Hits:

[Multimedia Develop] An_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:

[Program doc] fullSim

Description: This a simulator written in Tcl to simulate a network node carrying GSM and GPRS traffics with QoS mechanisms. The payload type including circuit-switched voice, VoIP and web traffic, and the performance including packet drop, delay can be analyzed. The implemented QoS mechanism is DiffServ, with 4 RED queues for different services with different priorities.-This is a simulator written in Tcl to simulate a network node carrying GSM and GPRS traffics with QoS mechanisms. The payload type including circuit-switched voice, VoIP and web traffic, and the performance including packet drop, delay can be analyzed. The implemented QoS mechanism is DiffServ, with 4 RED queues for different services with different priorities.
Platform: | Size: 26256384 | Author: wang haibo | Hits:

[Other] QualityofServiceProvisioning

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 stack] Phone

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:

Category

Source Code

Web/Internet

Develop Tools

Document

Other

Search in results

OS

Platform

Language

File Type

Search list