webrtc streaming

Transferred from: http://www.cnblogs.com/gbin1/archive/2013/03/26/2982917.htmlWEBRTC changed the network, it helped us to be impossible to achieve in a few months ago, even the things that we dare not think about become a reality. Whether you're making video chats by visiting URLs or sharing files on your social network, WEBRTC is rapidly expanding the application horizon and looking for what can be achieved in Web applications.WEBRTC is a recommended

Recently, our team is developing a app to help people solve problem face to face.We Choose WEBRTC Protocol as our bridge among different platform (Android, IOS, browser etc).But there are a hole in Android 6.0 system, the protocol can not support Android 6.0 system.As we known, Libjingle (Http://mvnrepository.com/artifact/io.pristine) was built in December, 2015,It hasn ' t been updated in least one year. I do not know if

WEBRTC 's echo cancellation (AEC, AECM) algorithm mainly includes the following important modules: Echo delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC algorithm should also include double-ended detection (DT). Considering that the NLMs, NLP and CNG used by WEBRTC belong to the classical algorithm

WEBRTC Compilation Details--record + reprintOriginal address: http://blog.csdn.net/temotemo/article/details/7056581WEBRTC compilingMy environment:Operating system: XP SP3VS 2013Tools required before compiling the source codeGet the source code tool:1, first need to install the source of the tool SVN (Project code version management tools, Google also use this)TortoiseSVN 1.6.12http://sourceforge.net/projects/tortoisesvn/2. Download and install Msysgit

WEBRTC in the Chrome browser demo Many examples, WebRTC source, but in the Firefox browser, the example can not be used, the information on the web said to set the media.peerconnection.enabled to True, However, in the Firefox browser, the default value is True, using the WEBRTC example in Firefox or can not capture local video. Here are the examples found online:

standard has gone into the tens of thousands of millions of ipads and iphones, making it the absolute overlord of the video coding field, with a share of over 80%. It is also widely used in the field of network streaming media data, various HDTV broadcasting and satellite TV broadcasting. Compared with the previous coding standards have some new features, such as multi-reference frame motion compensation, variable block size motion compensation, intr

From http://www.dataguru.cn/article-9532-1.html The demand for distributed streaming is increasing, including payment transactions, social networks, the Internet of Things (IoT), system monitoring, and more. There are several applicable frameworks for convection processing in the industry, so let's compare the similarities and differences of each stream processing framework.Distributed stream processing is the continuous processing, aggregation and a

WebRTC represents the best technology in the field of real-time communication since the date of birth. But for a long time, it supported the video encoder only VP8, and later with H265/VP9 as the representative of the birth of the next generation of video Encoders, WebRTC appeared VP9 Codec. The most widely used H264 has been kept out of sight. Until Cisco announces its H264 codec open source for OpenH264,

WEBRTC Echo Cancellation (AEC, AECM) algorithm introduction NBSP;WEBRTC echo Cancellation (AEC, AECM) algorithm mainly includes the following important modules: 1. Echo Delay Estimation 2.NLMS ( Normalized minimum mean square adaptive Algorithm) 3.NLP (nonlinear filtering) 4.CNG (Comfort Noise generation), the general classic AEC algorithm should also include double-ended detection (DT). Considering that

Welcome to Join WEBRTC Learning Group (659922087) to obtain free learning resources, mutual communication and growth. WEBRTC of the Echo Cancellation (AEC, AECM) algorithms mainly include the following important modules: Echo delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC algorithm should also inc

The previous article (latency) and the reduction method at the end of the speech communication, said that from the beginning of this article will cut into the WebRTC Neteq theme, Neteq is one of the two core technologies of audio technology in WEBRTC (another core technology is the front and back processing of audio, including AEC, ANS, AGC etc, commonly known as 3 a algorithm).

app Google launched), we'll see what chemistry can produce.So regardless of whether duo succeeds or not, at least we see Google's focus on social and video. In other words, even if duo is unsuccessful, Google will definitely launch other relevant apps to get into this area.2, Google is not always pushing the HTML5 standard? And there is a very important element in HTML5 is WEBRTC, on such an important occasion to show duo (Duo is based on

The goal of this project is to develop an online multiplayer game with as few server resources as possible, while expecting to run the game on one user's browser while allowing another player to connect. In addition, it is hoped that the program is as simple as possible for analysis in blogs.Application of the technologyI found WebRTC when I first came into contact with the technology, and I thought the technology was right for the project.

The newer WEBRTC source code does not have the corresponding vidoeengine with the voiceengine structure, instead of the meidaengine. Mediaengine includes the Mediaengineinterface interface and the fact that the compositemediaengine,compositemediaengine itself is also a template class, and two template references are audio engines and video engines respectively. The compositemediaengine derived class Webrtcmediaengine depends on the template parameters

?? Next Tuesday launch of "Audio and video technology WEBRTC" Open class, Welcome to join!Open Course Links: http://edu.csdn.net/huiyiCourse/detail/90The course's explanatory material and code will be uploaded at the end of the Open class http://download.csdn.net/user/yangzhenpingThe following is the course information:Course Brief IntroductionThe core of WEBRTC originates from Gips.Gips (Global IP Sound) w

For project reasons, audio and video is required, so the open source WebRTC (with BDS open source agreement) and Google supported open source project are selected. On the HTML5 side, Google's ambition is generally visible, why so, WEBRTC support browser. Currently support WebRTC Browser has Chrome,firefox,opera (the latest version, the old version is not supporte

This article mainly introduces WEBRTC in each platform debug or log viewing mode, to facilitate troubleshooting, including Bs,pc,android,ios (this series of articles reproduced please indicate the source, blog Park rtc.blacker).1, Browser development:This development method does not need to download and compile WEBRTC source code (many people are "dead" here, but it is really troublesome, the reason is not

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

 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