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
For project reasons, audio and video is required, so the open source WebRTC (with BDS open source agreement) and Google supported open source project are selected. On the HTML5 side, Google's ambition is generally visible, why so, WEBRTC support browser. Currently support WebRTC Browser has Chrome,firefox,opera (the latest version, the old version is not supporte
1,HTTP://WWW.WEBRTC.ORG/WEBRTC official website, god Horse compilation, God horse download, the solution here is the most authoritative.---------------------------------2,HTTPS://CODE.GOOGLE.COM/P/WEBRTC/WEBRTC Source download location, you can also pay attention to the latest changes anywhere.----------------------------------3,https://webrtchacks.com/is an arti
ObjectiveThe purpose of this article is to save you 10 hours (or more) of your life, or to waste 10 minutes. WEBRTC's compilation has been called a nightmare as a large cross-platform base library that Google has frequently updated. If you happen to want to compile WEBRTC under Windows, you'd better evaluate your patience in advance.As of now, I have tried almost all the articles in the Chinese blog community, which can be said to have failed. I do no
Ubuntu14.04 compile WebRTC For Android code 2014-07-24, ubuntu14.04webrtc
I haven't written a blog for almost a year. Recently, I developed an instant messaging project based on Google's open-source WebRTC project. During this project, I encountered some problems when downloading WebRTC code, this is a record here, and we hope to help kids who encounter similar
I have only recently started to study webrtc deeply. If I have any questions, please leave a message.How to generate WEBRTC vs engineering under Windows see my last article.But when I modify the project, such as adding cc and H files, adding a third-party dependency library, you will find that the VS modification is useless. VS can only be used when a code reader and editor are available.This time we need t
Turn from: http://blog.csdn.net/lixiaowei16/article/details/53407010
Audio and video synchronization is related to the most intuitive user experience of multimedia products, audio and video media data transmission and rendering playback of the most basic quality assurance. If the audio and video is not synchronized, it may cause delays, such as cotton, etc. very affect the user experience phenomenon. Therefore, it is very important. Generally speaking, the audio and video synchronization maint
The bandwidth assessment (BWE) is perhaps the most critical module in the WEBRTC video engine, which determines the amount of video data that can be generated when network congestion is not raised in video traffic.
Early bandwidth assessment algorithms are relatively primitive, mostly based on packet loss estimation, the basic strategy is to gradually increase the amount of data sent, until the loss of packets detected. In order for the sender to lear
, YY streaming media Services is basically this function, the structure is as follows.Advantages of the program:1.1, compared with mixer, the service-side pressure is relatively small. and easy to expand.1.2, low latency, especially in combination with SVC, can greatly improve the client experience.Disadvantages of the program:1.1, considering that different clients need different receiving ability, so the architecture of the server is really not simple.Additional notes:1, because the various mo
What is WEBRTC.
As we all know, the browser itself does not support each other directly to establish channels for communication, are through the server relay. For example, now there are 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 message sent to the server, the server to a message relay, sent to B, and vice versa. In this way a message between A
Transferred from: Http://www.oschina.net/p/kurentoKurento is a WebRTC streaming media server and some client APIs, which makes it easier to develop advanced video applications for WWW and smart phone platforms. The types of applications that can be developed using Kurento include video conferencing, audio and video broadcasting, audio and video recording, transco
Compile and install WebRTCsvncheckouthttp in Ubuntu: // configure. Then gclientconfighttps: // webrtc.
Compile and install WebRTC in Ubuntu
Svn checkout http://webrtc.googlecode.com/svn/trunk/
After the download is complete, WebRTC will get a folder named trunk by default, which contains the WebRTC source code, which i
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC Technology is committed to the browser to achieve real-time audio and video, multimedia data interoperability, its NAT traversal part of the ice framework, the purpose is to achieve media P2P,SBC called the session Border controller, dedicated to the media, signaling NAT traversal, but SBC technology in the media by the server relay, This violates the original intention of
side using normal HTTP methods
The client executes a JavaScript script in the Web page that establishes a connection to the server
Between the server and the client, you can send valid data to each other in two directions
The server can send data to the client in real time, and the client can send the data to the server in real time.
You need a server that can do event loops
Using WebSockets to allow connections to be established across domains
It also supports third-par
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.
I built a communication learning Exchange Group, 45211986, Welcome to join.
WEBRTC Technology is committed to the browser to achieve real-time audio and video, multimedia data interoperability, its NAT traversal part of the ice framework, the purpose is to achieve media P2P,SBC called the session Border controller, dedicated to the media, signaling NAT traversal, but
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.When using WEBRTC Base library to develop Android,ios native application, because the mobile terminal is not like PC side, in bandwidth stability, system performance is very big difference, so for mobile device WEBRTC need to do some optimization to improve the call effect,For example, WE
The online example of WEBRTC is mostly code, the following is an example of a WEBRTC-to-one sample of code simplification, which is tested under Chrome 37. Where iceserver can be omitted, there is no iceserver when the same LAN can still communicate.Client code:When WEBRTC is implemented, the signaling server is required to facilitate communication between client
Introduction APPRTC is what, webrtc.org official Experience App Ingredients: ubuntu14.04, other Linux versions are not limited, the official does not specify Chrome M51+stunnle Https://www.stunnel.org/ind Ex.htmlrfc5766-turn-server https://code.google.com/archive/p/rfc5766-turn-server/Google App Engine SDK for Pythonapp RTC HTTPS://GITHUB.COM/WEBRTC/APPRTCsteps:Set up proxyBecause of the particularity of the domestic network, this step is very importa
Noise suppression, is what everyone said noise reduction. This noise reduction is a distinction between vocals and non-human voices, which is a noise.A piece of audio that contains vocals and noise is processed by the module, and in theory, only the vocal is left.WEBRTC NS In the industry is still famous, through the actual comparison test, we found that webrtc noise reduction is indeed performance and stabilityare higher than similar open source algo
Recently work used to WEBRTC, found that WEBRTC is a treasure trove, there are many things worth studying.Search this area a lot of information, found that the introduction of the use of WEBRTC, but for some of the algorithm researchNot much. In particular, the algorithm can be said to be concise and clear is even rarer.In fact, I would like to carefully study ea
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