webrtc java

Transferred from: http://www.cnblogs.com/fangkm/p/4364553.htmlWEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether usi

WEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether using the

This paper focuses on the WEBRTC-based direct-to-peer streaming technology (Shi, Pro Gajun CTO, Editor: Dora), first published in " here "Support the original, reprint must indicate the source, welcome attention to the public number blacker (Id:blackerteam or WEBRTCORGCN)So far, the live industry continues as expected in full swing development, in the competition after the delay, HD, beauty, seconds open and other functions, the recent major live plat

 Harnessing Open Source Library WebRTC Fourth chapter-Compiling Macios edition Author: Adam Acknowledgements: Lao Zhang Date: 2015-4-6 Version: 1.0.0 Welcome reprint, has the question feedback q:2780113541, as far as possible consummates series of tutorials. Update Address: Https://github.com/wpc320/webrtc_doc.git Depot_tools proxy settings Reference old Zhang "the best wall in history download WEBRTC co

2016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchronized, it is possible to cause delay, lag, etc. that affect the user experience very mu

Recently flv.js things seem to have ignition, and again to the MSE this thing to bring up.MSE (Media source extensions) is a new function of HTML5, and the general function is to realize streaming media function.If the MSE with WEBRTC and JS binary processing, then you can implement the server to send video to one of the browser users, the browser users will then transfer video streaming to other users of the function. is a web-side in the Peer-to-pee

The newer WEBRTC source has no voiceengine structure corresponding to the vidoeengine, replaced by Meidaengine. Mediaengine contains the Mediaengineinterface interface and its implementation compositemediaengine,compositemediaengine itself is also a template class, two template parameters are the audio engine and video engine respectively. Compositemediaengine derived classes Webrtcmediaengine dependent template parameters are Webrtcvoiceengine and We

Compile and install WebRTCsvncheckouthttp in Ubuntu: // configure. Then gclientconfighttps: // webrtc. Compile and install WebRTC in Ubuntu Svn checkout http://webrtc.googlecode.com/svn/trunk/ After the download is complete, WebRTC will get a folder named trunk by default, which contains the WebRTC source code, which i

Google's first integration of WebRTC in the Chrome Dev release released this January was a source of widespread concern. Today, Google published a roadmap for the development of WebRTC technology in its blog.WebRTC is a technology for real-time video and audio communication inside the browser, and Google acquired a technology in 2010 to acquire Global IP Solutions. The technology is based on the WHATWG prot

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC Technology is committed to the browser to achieve real-time audio and video, multimedia data interoperability, its NAT traversal part of the ice framework, the purpose is to achieve media P2P,SBC called the session Border controller, dedicated to the media, signaling NAT traversal, but SBC technology in the media by the server relay, This violates the original intention of

A recent study on how WebRTC speech runs on iOS found that the voice_engine of WebRTC has implemented iOS-related classes, but encountered a series of problems in specific applications. After several days of hard work, finally, we solved a series of problems and successfully realized recording and playing local loop in the simulator. Compile the testProgramIn the process, we plan to use the libjingle Libr

I have only recently started to study webrtc deeply. If I have any questions, please leave a message.How to generate WEBRTC vs engineering under Windows see my last article.But when I modify the project, such as adding cc and H files, adding a third-party dependency library, you will find that the VS modification is useless. VS can only be used when a code reader and editor are available.This time we need t

Turn from: http://blog.csdn.net/lixiaowei16/article/details/53407010 Audio and video synchronization is related to the most intuitive user experience of multimedia products, audio and video media data transmission and rendering playback of the most basic quality assurance. If the audio and video is not synchronized, it may cause delays, such as cotton, etc. very affect the user experience phenomenon. Therefore, it is very important. Generally speaking, the audio and video synchronization maint

The bandwidth assessment (BWE) is perhaps the most critical module in the WEBRTC video engine, which determines the amount of video data that can be generated when network congestion is not raised in video traffic. Early bandwidth assessment algorithms are relatively primitive, mostly based on packet loss estimation, the basic strategy is to gradually increase the amount of data sent, until the loss of packets detected. In order for the sender to lear

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) In the previous article, we explained WebRTC 's overview, history, security, and developer tools. The next step is to explain the process of building

Since the beginning of WEBRTC development, often asked by others, Safari browser can support WEBRTC? I also want Safari to support WEBRTC so you don't have to write native WebRTC apps or Safari plugins.Happily, Apple recently announced that WEBRTC will enter Safari and iO

Reproduced in the original: http://blog.csdn.net/shichaog/article/details/52399354 thank you very much. Vad (voice Activity Detection) algorithm is to detect the voice , in the far-field speech interaction scenario, VAD faces two challenges: 1. How to successfully detect the lowest energy voice (sensitivity).2. How to successfully detect in the multi-noise environment (detection rate and false detection rate).The missed response is originally the voice but not detected, and the virtual detection

http://blog.csdn.net/nonmarking/article/details/47910043 This series is currently a total of three articles, follow up will also update WebRTC Videoengine Ultra-Detailed tutorial (i)--the basic process of video Call WebRTC Videoengine Ultra-Detailed tutorial (ii)--integrated OPENH264 codec WebRTC Videoengine Ultra-Detailed tutorial (iii)-- Overview of Integrate

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.When using WEBRTC Base library to develop Android,ios native application, because the mobile terminal is not like PC side, in bandwidth stability, system performance is very big difference, so for mobile device WEBRTC need to do some optimization to improve the call effect,For example, WE