angular webrtc

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4374610.htmlThe first two articles describe the running process of WEBRTC and the use of the framework interface, and then begin to analyze the local audio and video collection process. Due to the large space, video capture and audio capture are divided into two blog posts, where the video capture process is analyzed first. Analysis of the time of the first analysis of the

src\webrtc\examples\peerconnection\client\conductor.cc1, BOOL Conductor::initializepeerconnection ()1.1 WEBRTC::createpeerconnectionfactory();src\talk\app\webrtc\peerconnectionfactory.cc2, BOOL Peerconnectionfactory::initialize ()2.1.1 cricket::mediaengineinterface* peerconnectionfactory::createmediaengine_w() {Return Cricket::webrtcmediaenginefactory::create(Def

Transferred from: http://blog.csdn.net/nonmarking/article/details/47375849 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)--integr

Cross-platform WEBRTC WEBRTC is Google Open source of a plug-in real-time video communication technology, which is divided into web development and native development; currently supports Chrome,firefox,android,ios,opera,edge. is a true sense of cross-platform plug-in real-time video communication technology. Video applications are generally based on web-level development. This paper is mainly about the cod

In "Let WEBRTC support H264 codec" I provide a priority to use the H264 codec thinking. We can verify it on the browser side. There are three ways to verify: In JS print SDP view Chrome's log chrome_debug.log (see Open Chrome Log) Grab bag using webrtc-internals The first three kinds are no longer introduced, we look at the webrtc-internals. The

Angular (3) of the front-end framework and angular of the front-end framework This section describes Angular's dirty checking, functions, and services.1. Dirty checking Angular usesdirty checkTo track data changes, which is the basis of two-way data binding. The so-called dirty check means angular binds a watcher to ea

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

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 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,

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

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

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: