webrtc h264

Www. fire-rain.comblogFFMPEG_H264_MP4PHP + FFMPEG automatic transcoding H264 standard Mp4 File recently made an online teaching network project, need to upload any format of video automatic h264 standard video, using html5 playback. Finally, PHP + FFMPEG is used for implementation. Here we will share our detailed solutions with you! Configure ph Http://www.fire-rain.com/blog/FFMPEG_H264_MP4 PHP + FFMPEG aut

A recent study on how WebRTC speech runs on iOS found that the voice_engine of WebRTC has implemented iOS-related classes, but encountered a series of problems in specific applications. After several days of hard work, finally, we solved a series of problems and successfully realized recording and playing local loop in the simulator. Compile the testProgramIn the process, we plan to use the libjingle Libr

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

Recently flv.js things seem to have ignition, and again to the MSE this thing to bring up.MSE (Media source extensions) is a new function of HTML5, and the general function is to realize streaming media function.If the MSE with WEBRTC and JS binary processing, then you can implement the server to send video to one of the browser users, the browser users will then transfer video streaming to other users of the function. is a web-side in the Peer-to-pee

The newer WEBRTC source has no voiceengine structure corresponding to the vidoeengine, replaced by Meidaengine. Mediaengine contains the Mediaengineinterface interface and its implementation compositemediaengine,compositemediaengine itself is also a template class, two template parameters are the audio engine and video engine respectively. Compositemediaengine derived classes Webrtcmediaengine dependent template parameters are Webrtcvoiceengine and We

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

reproduced in the original: http://www.cnblogs.com/lidabo/p/6842765.html thank you very much. "WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is a technology that supports web browsers for real-time voice conversations or video conversations, and is Google's $68.2 million acquisition of Global in 2010 IP Solutions A technology that Google has made available to the company. ”

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