webrtc h264

app Google launched), we'll see what chemistry can produce.So regardless of whether duo succeeds or not, at least we see Google's focus on social and video. In other words, even if duo is unsuccessful, Google will definitely launch other relevant apps to get into this area.2, Google is not always pushing the HTML5 standard? And there is a very important element in HTML5 is WEBRTC, on such an important occasion to show duo (Duo is based on

The goal of this project is to develop an online multiplayer game with as few server resources as possible, while expecting to run the game on one user's browser while allowing another player to connect. In addition, it is hoped that the program is as simple as possible for analysis in blogs.Application of the technologyI found WebRTC when I first came into contact with the technology, and I thought the technology was right for the project.

The newer WEBRTC source code does not have the corresponding vidoeengine with the voiceengine structure, instead of the meidaengine. Mediaengine includes the Mediaengineinterface interface and the fact that the compositemediaengine,compositemediaengine itself is also a template class, and two template references are audio engines and video engines respectively. The compositemediaengine derived class Webrtcmediaengine depends on the template parameters

?? Next Tuesday launch of "Audio and video technology WEBRTC" Open class, Welcome to join!Open Course Links: http://edu.csdn.net/huiyiCourse/detail/90The course's explanatory material and code will be uploaded at the end of the Open class http://download.csdn.net/user/yangzhenpingThe following is the course information:Course Brief IntroductionThe core of WEBRTC originates from Gips.Gips (Global IP Sound) w

The recent need to encode H264 video into MP4 format. Research, one method is to use the FFmpeg library, you can first decode the H264 file, and then encode the generation of MP4 files, but this method is less efficient, 10M video may take a few seconds to complete. Another way is to encapsulate the H264 package directly into the MP4 format according to the MP4 f

Reprinted from: http://hi.baidu.com/y11022053/item/6d4c34ba87c7b5f362388e9a After installing the FFmpeg, if you use the FFmpeg tool to turn a video file into H264 video encoding, MP3 audio encoding or other ffmpeg itself without the XXX encoding type, you will see the error message, unknown encoder ' xxx '. Now all you need is to install the other encoder on the line, essentially the other encoder in the form of a library installation, for example, t

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4374610.htmlThe first two articles describe the running process of WEBRTC and the use of the framework interface, and then begin to analyze the local audio and video collection process. Due to the large space, video capture and audio capture are divided into two blog posts, where the video capture process is analyzed first. Analysis of the time of the first analysis of the

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

Transferred from: http://blog.csdn.net/nonmarking/article/details/47375849 This series is currently a total of three articles, follow up will also update WebRTC Videoengine Ultra-Detailed tutorial (i)--the basic process of video Call WebRTC Videoengine Ultra-Detailed tutorial (ii)--integrated OPENH264 codec WebRTC Videoengine Ultra-Detailed tutorial (iii)--integr

RTP (H264) First package (single NAL unit mode) ————-SPS 80{v=10,p=0,x=0,cc=0000}60{m=0,pt=110 0000}5370{Sequence Number}9A FC AF27{timestamp}69 25 77 {ssrc} 67 4 d 40 33 54 0c 04 B4 20 00 00 03 00 00 00 0 c D1 E3 06 54 (SPS H264) RTP (H264) second package (single NAL unit mode) ——— PPS 80 {V=10,p=0,x=0,cc=0000} 60 {m=0,pt=110 0000} 53 71{sequenc

Ubuntu14.04 install GStreamer to test UVC H264 CameraRecently debugging UVC H264 Camera, need to test the camera in Ubuntu, so with the help of GStreamer to achieve.Gtreamer is a programming framework based on the glib library (the latest version needs glib2.0) to build streaming media applications, the goal of which is to simplify the development of audio/video applications that can now be used to deal wit

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

This article mainly introduces WEBRTC in each platform debug or log viewing mode, to facilitate troubleshooting, including Bs,pc,android,ios (this series of articles reproduced please indicate the source, blog Park rtc.blacker).1, Browser development:This development method does not need to download and compile WEBRTC source code (many people are "dead" here, but it is really troublesome, the reason is not

Transferred from: http://www.cnblogs.com/fangkm/p/4364553.htmlWEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether usi

WEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether using the

 Harnessing Open Source Library WebRTC Fourth chapter-Compiling Macios edition Author: Adam Acknowledgements: Lao Zhang Date: 2015-4-6 Version: 1.0.0 Welcome reprint, has the question feedback q:2780113541, as far as possible consummates series of tutorials. Update Address: Https://github.com/wpc320/webrtc_doc.git Depot_tools proxy settings Reference old Zhang "the best wall in history download WEBRTC co

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

WEBRTC 's echo cancellation (AEC, AECM) algorithm mainly includes the following important modules: Echo delay estimation, NLMS (normalized minimum mean square adaptive algorithm), NLP (nonlinear filtering), CNG (Comfort noise generation). The General classic AEC algorithm should also include double-ended detection (DT). Considering that the NLMs, NLP and CNG used by WEBRTC belong to the classical algorithm

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

Options rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 1User-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 1Date: sat, Jan 01 2000 00:05:11 GMTPublic: Options, describe, setup, teardown, play, pauseDescribe rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 2Accept: Application/SDPUser-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 2Date: