pusher webrtc

src\webrtc\examples\peerconnection\client\conductor.ccboolconductor::initializepeerconnection()1 webrtc::createpeerconnectionfactory ();src\talk\app\webrtc\peerconnectionfactory.cc1.1 New Rtc::refcountedobject1.2 bool Peerconnectionfactory::initialize ()1.2.1 cricket::mediaengineinterface* media_engine =Peerconnectionfactory::createmediaengine_w()src\talk\media\

WEBRTC Technology Group: 234795279Original Address: http://blog.csdn.net/temotemo/article/details/7530504 1, WebRTC purpose WebRTC (Web real-time communication) The ultimate purpose of the project The main is to allow web developers to be based on the browser (chrome\firefox\ ... Fast and easy to develop rich real-time multimedia applications, without the need to

based on 3GPP Ims/rcs and can be used in embedded and desktop systems. The framework is written using ansci-c and is very portable. and has been designed to be very lightweight and effective in embedded systems with low memory and low processing power. The Idoubs feature on the Apple system is based on this framework. Most of the audio and video encoding formats are supported (H264 (VIDEO), VP8 (video), ILBC (audio), pcma,pcmu,g722,g729). NAT supports Ice (Stun+turn)2) Effect measurementTest en

Recently experiment how to let WEBRTC support H264 code, record, for people who need reference. To illustrate, I was compiling the WebRTC under Ubuntu Server 14.04, using the native (c + +) API to develop WebRTC applications. So my adjustments are based on the native code. The end result is that the browser can send a video with H264 or receive H264 video. Note t

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

WEBRTC reply content: I am in development and have a basic understanding of the WebRTC source code stack. It mainly consists of two key technologies: 1. webRTC Video/Voice Engine, including camera microphone operations, Video preprocessing, VP8 coding/decoding, and streaming media transmission (RTP/RTCP); 2. implement the P2P channel and use libjingle to complete

Original address: http://segmentfault.com/a/1190000000436544 what is WEBRTC?It is well known that the browser itself does not support the direct establishment of communication channels between each other through the server to relay. For example, there are now two clients, A and B, they want to communicate, first need a and server, B and server to establish a channel between. A to send a message to B, a first send the message to the server, the server

Recently experiment how to let WEBRTC support H264 code, record, for people who need reference.To illustrate, I was compiling the WebRTC under Ubuntu Server 14.04, using the native (c + +) API to develop WebRTC applications. So my adjustments are based on the native code.The end result is that the browser can send a video with H264 or receive H264 video.Note that

#!/bin/bashfunction Build_xcode () {Echo "* * * Building WebRTC for the ia32 IOS simulator";Export gyp_generators= "xcode";Export gyp_defines= "build_with_libjingle=1 build_with_chromium=0 libjingle_objc=1 os=ios target_arch=ia32 clang_ Xcode=1 ";Export gyp_generator_flags= "$GYP _generator_flags output_dir=out_ios_ia32";Export gyp_crosscompile=1;Gclient runhooks;Ninja-c Out_ios_ia32/release-iphonesimulator Iossim apprtcdemo;}function Build_iossim_ia3

ObjectiveBefore writing the article "separate compilation using WEBRTC audio processing module-Android", I have been trying to compile the WEBRTC audio processing engine Voe the whole to use for his project, but limited to the poor technology, time is tight, so did not succeed. The AECM, AGC, NS, and VAD modules in the engine were compiled separately to make it work. Although it can achieve a certain effect

Compile WebRTC For Android code in Ubuntu 14.04 Recently, a real-time communication project for audio and video chats was developed based on Google's open-source WebRTC project. Some problems were encountered during the download of WebRTC code, which was recorded here, we also hope to help the children's shoes who encounter similar problems while downloading and

Said Nat before the penetration of a few about the concept of WEBRTC, may have been the same as the author of the WEBRTC concept of the wrong understanding of the classmate. WebRTC (Network real-time communication) It is a Web browser to support real-time voice dialogue or video dialogue technology, it provides us with video conferencing core technology, includin

The greatest feature of real-time streaming media applications is real-time, while latency is the biggest enemy of real-time sex. The processing speed of media data is the important reason of delay, and the network congestion is the main cause of delay from the point of transmission. Network congestion can cause packet loss, and may result in longer data transfer times and increased latency.Congestion control is one of the important methods in real-time streaming media application quality assura

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) In this article we continue to learn about WebRTC 's related Api,rtcpeerconnectiont and Rtcdatachannel.RtcpeerconnectionRtcpeerconnection is a

reproduced in the original: http://blog.csdn.net/u012931018/article/details/17045077 thank Bo Master. WEBRTC Echo Cancellation (Acoustic ECHOCANCELLATION,AEC, acoustic echocancellation for MOBILE,AECM) algorithm mainly includes the following important modules: echo Time delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC

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:

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