I'm trying to study webRTC and want to learn about how it encodes and transport the media stream.
Are codes of this part of webRTC already included in the chromium source codes?
http://code.google.com/p/chromium/wiki/LinuxBuildInstructions
Thanks~!
yes and no.
The core of webrtc (including all the transport) is in the third party libwertc code, you can find information at webrtc.org.
The codec and other libraries are shared with Chrome, and libwebrtc import a copy of chromium when building from source. THe exact list of libs that are shared are in this file
The "capturer" is in the chrome code, more info there.
Related
We compiled WebRTC based on souce code ,using code of milesone m91,and with sctp not built.
1.Our code is based at WebRTC m91
2.usrsctp is disalbed and not built by setting these values to false in WebRTC.gni,so that usrsctp won't be built.
1) rtc_enable_sctp=false
2) rtc_build_usrsctp=false
3.build libjingle_peerconnection_so.so by command
autoninja -C out/arm libjingle_peerconnection_so
4.integrate the compilied libjingle_peerconnection_so.so to App and submit it to google play store
Though usrsctp is not built,however,the google play store detects that I'm using a vulnerable versions of WebRTC use usrsctp.
Here is the FAQ from google play store.
How to fix apps with bad WebRTC versions
Here is my questions
Question1：
How does google store dectect that I'm using a vulnerable versions of WebRTC use usrsctp ?The rules are not clear to me .
by scanning the symbols in .so ?
by scanning the meta infos ?
Question2:
Is there a way to know the version of WebRTC that libjingle_peerconnection_so.so is built from ?
In other words,given a specific so file libjingle_peerconnection_so.so ,How can I know that it's built from M91 or M102 ?Is there any meta informations about the version of WebRTC we are used in the compiled products ?
WebRTC M012 or higher is required to pass this check on Play, even if usrsctp is removed. While usrsctp is our most pressing concern, many other security issues have been fixed in other components. It is highly recommended that you update to a more recent version, and be prepared to update on an on-going basis in the future if more serious vulnerabilities.
Please don't attempt to circumvent this check in Play, as you will miss important notifications about future security issues, and there is no guarantee that detection will remain the same
I've filed an issue to WebRTC Issue 14664: usrsctp is not built but detected by goolge play store
I'm making Automated Zoom Opener for Zoom online class.
But, I don't know how to open website with Kotlin/Native on Windows PC.
So I searched for Kotlin module which is similar to webbrowser module in python.
I found a module, kotlinx.browser, but I couldn't find any information about it.
Can you find or provide some information and sample code about it?
(Or other ways to open website with Kotlin/Native on Windows PC.)
The package kotlinx.browser is part of kotlin-stdlib. According to (1) it needs Kotlin/JS as prerequisite. See (2) how to set up Kotlin/JS. A small example on the usage of kotlin.browser can be found at (3).
I am trying to add additional codec support to Chromium for HTML5 media play back. I've installed the package chromium-codecs-ffmpeg-extras to include a few more than what is included in original download.
I am curious if anyone knows the amount of work/ if its possible to fork the chromium code and add additional codec supports (.ts, .avi, etcc). Is it as simple as forcing chromium to use a different ffmpeg library or possibly use VLC to decode all of its videos?
I've got a simple application using WebKitGTK 1.6.0 that simply displays a webpage in a window.
When I browse a page using it that contains an Ogg/Theora video and HTML5 video element it appears this simple application is using some mozilla plugins and even xulrunner. Can somebody tell me why this is?
Here is a list of files open by the program using lsof.
I built WebKitGTK 1.6.0 and most of it's dependencies from source including gstreamer-0.10 and the gstreamer plugins. No where do I recall configuring WebKitGTK to using any mozilla plugins or xulrunner.
What might be causing the mozilla plugins, libtotem and xulrunner to be loaded? I provided the link to the files the application is using in hopes that it might provide a clue.
Lol, found the reason. I have FireFox, totem and totem-mozplugin installed on my system. Once I removed totem and totem-mozplugin, WebKitGTK used the appropriate Gstreamer plugins to render the video and audio.
Hi could anyone to tell me how to install depot tools.I went through documentation and trying to install the webrtc in windows,I installed visual studio 2010 and some related stuff given in documentation still i am in confusion and many questions like why visual studio is required? are we have write code in visual studio?It acts like IDE for webrtc?
If not please tell me where we have to place our project in local file system and how to implement my first demo application..I found a video published by google demonstrating about webrtc in that they wrote code in a html file so is html file itself enough to develop.?
Though i have done a sample application in a html file still i am getting blank page and in console i am getting as UNCAUGHT EXCEPTION near navigator.webkitGetUserMedia() please help me out to know more about webrtc.
I suggest you start by watching the excellent videos about WebRTC from Justin Uberti and Cullen Jennings, and then (shameless self promotion) get to grips with the code and examples in Getting Started With WebRTC on HTML5 Rocks.
webrtc.org is the home of the WebRTC project -- lots of resources and demos there.
Just last month we released an open source project aimed to get developers into WebRTC fast and easy.
http://www.easyrtc.com/
It includes a cross-platform server kit which will run in Windows. Working demos are included.
I just built the webrtc solution using VS2010 by following this web pages instructions and NOT using cygwin. I had but one issue which to resolve required the manual copying of two files into the correct folder, you will know which two files should you see the names of these files and read the error, you will then know the expected location when the build fails.
Ultimately you will get a peerconnection_server.exe and a peerconnection_client.exe by way of sample. I was able to modify them easily.
Hope this helps. Be sure to install the prerequisites and follow the instructions to the letter.
As a clarification, the webrtc C++ library is one implementation you can use for the protocol. Another is using the webrtc implementation in a (very) modern web browser, such as Firefox or Chrome. Then, you'll just need HTML + JS, as per the articles in HTML5 Rocks.