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网络音视频通话资料  

2009-09-28 10:32:42|  分类: 程序设计 |  标签: |举报 |字号 订阅

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libjingle  ---GTalk 使用的库

http://code.google.com/intl/zh-CN/apis/talk/libjingle/index.html


libjingle is a collection of open-source C++ code and sample applications that enables you to build a peer-to-peer application. The code handles creating a network connection (through NAT and firewall devices, relay servers, and proxies), negotiating session details (codecs, formats, etc.), and exchanging data. It also provides helper tasks such as parsing XML, and handling network proxies. You could build the following programs with libjingle:

  • A multi-user voice chat application
  • A multi-user video conferencing application
  • A multi-user live music streaming application
  • A peer-to-peer file sharing application

libjingle is available on Google Code for both Windows and UNIX/Linux operating systems. This documentation supplements the readme files included with the code.

We are providing this source code as part of our ongoing commitment to promoting consumer choice and interoperability in Internet-based real-time-communications. This code is made available under a Berkeley-style license, which means you are free to incorporate it into commercial and non-commercial software and distribute it.

Start with the Developer Guide page to learn how to use libjingle.

Linphone--- 一个开源网络电话项目 http://www.linphone.org/index.php/eng/features

List of supported features:

  • audio with the following codecs: speex (narrow band and wideband), G711 (ulaw,alaw), GSM, and iLBC (through an optional plugin)
  • video with codecs: H263-1998, MPEG4, theora and H264 (thanks to a plugin based on x264), with resolutions from QCIF(176x144) to SVGA(800x600) provided that network bandwidth and cpu power are sufficient.
  • Supports any webcam with V4L or V4L2 driver under linux
  • Any webcam on windows
  • text instant messaging and presence (using the SIMPLE standart)
  • Addressbook
  • DTMF (telephone tones) support using SIP INFO or RFC2833
  • understands SIP ENUMS (sip phone numbers using the naptr DNS service, without proxy)
  • echo cancelation using the great speex echo canceler
  • SIP proxy support: registrar, proxies, with digest authentication
  • STUN support for traversal of UDP NATs (=firewall)
  • sound backend using either ALSA (the most efficient), OSS, or arts (kde sound daemon)
  • supports IPv6
  • the bandwidth limitations are signaled using SDP (b=AS...), resulting in audio and video session established with bitrates that fits the user's network capabilities.
  • can use plugins: to add new codecs, or new core functionalities, such as remote directory search of sip addresses for example.

Speex: A Free Codec For Free Speech

http://speex.org/   一个很强大的语音编码库,基于人声处理做了很多压缩,GNU 项目来的

Overview

Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications by providing a free alternative to expensive proprietary speech codecs. Moreover, Speex is well-adapted to Internet applications and provides useful features that are not present in most other codecs. Finally, Speex is part of the GNU Project and is available under the revised BSD license.

The Technology

Speex is based on CELP and is designed to compress voice at bitrates ranging from 2 to 44 kbps. Some of Speex's features include:

  • Narrowband (8 kHz), wideband (16 kHz), and ultra-wideband (32 kHz) compression in the same bitstream
  • Intensity stereo encoding
  • Packet loss concealment
  • Variable bitrate operation (VBR)
  • Voice Activity Detection (VAD)
  • Discontinuous Transmission (DTX)
  • Fixed-point port
  • Acoustic echo canceller
  • Noise suppression

Note that Speex has a number of features that are not present in other codecs, such as intensity stereo encoding, integration of multiple sampling rates in the same bitstream (embedded coding), and a VBR mode; see our comparison page for more.



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