chromium webrtc

Chromium gesture plug-in Environment and symptoms Chromium 18.0.1025.151 (developer build 130497 Linux) Ubuntu 12.04 Many gesture plug-ins have been tested. They are displayed in the right-click menu and are not easy to use. Useful plug-ins The plug-in is smooth-gestures. Unfortunately, this plug-in has been put off the shelf because it collects privacy and does not repent. Google finds the following

This problem has troubled me for one hours ah, Baidu everywhere can not get results Ah! (This is Goole disabled to helpless Ah!) ）Before we solve the problem, we will comb the two points of knowledge.1.adobe Flash support for Linux system final version is 11.2, will not provide a high version later, but will be safe to keep the update. The following is the official website explanation. http://get.adobe.com/cn/flashplayer/Note:Adobe Flash Player 11.2 will be the last version that supports the Lin

Environment Description:System: CentOS 6.5 X64Very simple, the main reason is to start the chromium when the own flash is specified, we can be in the boot parameters in addition to the specified flash!sudo vim/usr/bin/chromium-browserchromium_rhel_flags= "--enable-plugins--enable-extensions--enable-user-scripts--enable-printing--enable-sync-- Auto-ssl-client-auth--ppapi-flash-path=/opt/

The previous blog post has been introduced through the prefetch pre-load Web pages of resources to improve page loading speed, let's take a look at the page acceleration of the chromium prerendering. Before introducing prerendering, two concepts are introduced:1.PLT (page load time) : The user browses the Web page to trigger the browser kernel onpagestarted to onpagefinished that is when the page starts to finish. 2.pplt (Perceived page load time): Us

When deploying WEBRTC or SIP-to-peer scenarios, you will often encounter environments that are not penetrated by peerThis is where the tunserver comes in.Here we use turnserver-0.7.3Download confuse dependent librarieswget http://savannah.nongnu.org/download/confuse/confuse-2.7.tar.gzTar zxvf confuse-2.7.tar.gzCD confuse*./configureMake make installDownloadwget http://downloads.sourceforge.net/project/turnserver/turnserver-0.7.3.tar.bz2Tar jxvf turns

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version of TELEMCU adds Android phone-side WEBRTC video conferencing capabilities, Android phone installs Chrome browser after loading TELEMCU WEBRTC client TeleWeb can directly participate in video conferencing,At the same time, TeleWeb can support two WEBRTC client-to-

Transferred from: http://www.cnblogs.com/lingyunhu/p/4232348.htmlToss for one weeks to finally set up the environment of Kurento (development stage of the product, some bugs to solve their own), so write a separate document to introduce.The following starts to introduce Kurento, the article from the blog Garden Rtc.blacker, reproduced please explain the source.What is Kurento?Video conferencing involves a one-to-many, many-to-many, broadcast, transcoding, mixing, co-screen, recording, which requ

WebRTC Point-to-point video calling system Main functions:1, based on the WebSocket online user list;2, use WebSocket as signaling channel, build WEBRTC video call.Github:https://github.com/graceup/webrtcDevelopment Ide:myeclipse 8.6 Engineering Code: UTF-8Environmental requirements: 1, TOMCAT requires more than 7.0 of the versionNote: When deploying, you need to change "ws://localhost:8080/" in the Js/con

WebRTC really is not very good to get, currently only the PC-side web page and mobile phone-side web page video. But there are still some problems. 1, both must use Firefox 2, feel pc-side camera shot out of the screen can also, the phone side a little bit of spending 3, enter the room after a period of time to show two video ~~~~APPRTC demo has not been tuned, the problem in Turnserver , and then sent the article. There are a lot of APPRTC on the Int

Google's Turnserver download method:svn checkout http://rfc5766-turn-server.googlecode.com/svn/branches/v3.2/ Rfc5766-turn-server-read-onlyAbout the application of WEBRTC Google gives an example:https://apprtc.appspot.com/(need FQ, sometimes fq may not be able to land, it is estimated that the use of too many people)I was always curious about the way he used turn, and then finally figured out what was going on. Take a look at the following characters:

H264 code Stream parsing, online has a lot of open source files; The general analysis is: Obtain Nalu,sps,pps,nalu type,slice type, obtain QP and so on; The computation can be obtained by the bitwise operation of C + +, but the structure can be obtained directly. Here is the WEBRTC in the H264 parsing Related: In the WEBRTC, about the H264 related source files in: webrtc58\src\

reproduced in the original: http://www.cnblogs.com/mod109/p/5469799.html thank you very much. The WEBRTC audio processing module is divided into noise reduction ns, Echo cancellation AEC(Echo control Acem), Automatic control gain AGC, Mute detection section. In addition WEBRTC has encapsulated a set of audio processing module APM, if it is not a special need, if users want to use the echo cancellation and

It is not recommended to compile individual modules in the WEBRTC separately for use. Yesterday, I was fortunate enough to ask the Google forum about the delay in computing the AECM module, and Project member said churn this delay actually didn't help the AECM effect, which only sped up the convergence of the built-in latency estimator when the AECM started, and if the delay in the update was incorrect, it would even make AE The CM built-in delay

Ubuntu under the Chromium browser is not supported by default Flash, in the new German software package search Flash get "Adobe Flash Player Plugin installer" and no egg, Because after the loading did not know how to run AH ~ Fall ~super Search Not! Execute sudo flashplugin-installer inside the terminal and say no command is found ~Workaround:Install Pepperflashplugin-nonfree (Note: It is really free). Either by new or apt-get.For an illustrated tutor

Import Collections Import struct import sys def ReadFile (filename, encoding): mode = ' RB ' if encoding = 0 Else ' RU ' W ITH open (filename, mode) as F:data = F.read () if encoding not in (0, 1): data = Data.decode (encoding) return Data Pack_file_version = 4 Header_length = 2 * 4 + 1 # two uint32s. (file version, number of entries) and # One uint8 (encoding of text resources) def Unpackdatapa CK (input_file): "" "reads a data pack file and returns a dictionary." " data = ReadFile (input_f

Recently doing a mobile end with mobile, web-side text, video, voice chat features. Text chat using WebSocket, a lot of information on the Internet, there is no difficulty. But in the video, voice chat encountered a small difficulty. have been looking for some of the SDK to quickly develop, such as Opentok, cloud communications, etc., but the project is used in the intranet, these SDKs must be used in an external network, you need to obtain signaling on their servers. Later, I will try to use

Crosswalk QuickStart, using WEBRTC (HTML) to start developing video callsInstall PythonDownload the installer from http://www.python.org/downloads/After the installation is complete, add the environment variable again.Installing Oracle JDK Download page:http://www.oracle.com/technetwork/java/javase/downloads/Select the Java version to download (recommended Java 7). Select a JDK to download and accept the license agreement. Once downlo

Some time ago in the audio and video version of iOS, so the title changed to Android IOS WebRTC audio and Video development summary, the following summarizes some of the experience in the development process:1. iOS WEBRTC audio and video compilation and download: have android WebRTC compile download experience and then go to get IOS, you will find a lot easier, a

Some time ago in the audio and video version of iOS, so the title changed to the Android iOS WEBRTC audio and Video development summary, the following summary of some experience in the development process: 1. iOS WEBRTC audio and video compilation and download: Have the android WEBRTC compile download experience to get IOS, you will find that more simple, then th

//existing Wm_timer If any would fire when delayed work should run. We//don ' t want to disturb that timer if it's already in flight. However,//If we did do all remaining delayed work and then lets kill the Wm_timer. if(More_work_is_plausible Delayed_work_time_.is_null ()) KillTimer (Message_hwnd_, reinterpret_cast This)); if(state_->should_quit) Break; if(more_work_is_plausible)Continue; More_work_is_plausible= state_->Delegate-doidlework (); if(state_->should_quit) Break; if(more_work_is_pl