webrtc windows

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

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

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

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

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

#!/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

WEBRTC Technology Group: 234795279 1. Voiceengine CODEC data structure WEBRTC, a struct struct codecinst is used to represent a specific audio codec object: struct Codecinst { int pltype; Payload Type Payload char plname[32];//payload name payload, 32 characters representing int plfreq; Payload frequence Load Frequency int pacsize; Packet size package int channels; Chan

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

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:

General Statement In the previous article, we explained how to integrate the OPENH264 codec into the WEBRTC, but OPENH264 can only encode baseline H264 video, and in terms of encoding quality, X264 is the best, This article will explain how to integrate the X264 encoder into the WEBRTC, in order to achieve decoding, at the same time to use the ffmpeg. The overall process, as before, is divided into the re-

Recently, due to the needs of the project, I began to touch the WEBRTC thing. Unexpectedly the threshold of this thing is still pretty high, next share I stepped on the pit, hoping for the first contact with this thing in the future to help people.WEBRTC official websiteThe first step of course is to see the official homepage (www.webrtc.org), first the content of the homepage was roughly swept over, probably a little bit of understanding of this thin

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 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