General Statement
In the previous article, we explained how to integrate the OPENH264 codec into the WEBRTC, but OPENH264 can only encode baseline H264 video, and in terms of encoding quality, X264 is the best, This article will explain how to integrate the X264 encoder into the WEBRTC, in order to achieve decoding, at the same time to use the ffmpeg. The overall process, as before, is divided into the re-
The main reference in this document is [1], which takes the code from the reference article. But [1] did not upload the complete code. Environment configuration can refer to the previous article [2]
The main realization of the video in the local transmission. It took a day.
As for WEBRTC video capture, codec, please refer to the online blog, here is not mentioned.
Compile using CMake to generate makefile, step cmake. Note the points that follow, an
Transferred from: http://www.cnblogs.com/gbin1/archive/2013/03/26/2982917.htmlWEBRTC changed the network, it helped us to be impossible to achieve in a few months ago, even the things that we dare not think about become a reality. Whether you're making video chats by visiting URLs or sharing files on your social network, WEBRTC is rapidly expanding the application horizon and looking for what can be achieved in Web applications.WEBRTC is a recommended
WEBRTC Echo Cancellation (AEC, AECM) algorithm introduction NBSP;WEBRTC echo Cancellation (AEC, AECM) algorithm mainly includes the following important modules: 1. Echo Delay Estimation 2.NLMS ( Normalized minimum mean square adaptive Algorithm) 3.NLP (nonlinear filtering) 4.CNG (Comfort Noise generation), the general classic AEC algorithm should also include double-ended detection (DT). Considering that
Welcome to Join WEBRTC Learning Group (659922087) to obtain free learning resources, mutual communication and growth. WEBRTC of the Echo Cancellation (AEC, AECM) algorithms mainly include 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 inc
The previous article (latency) and the reduction method at the end of the speech communication, said that from the beginning of this article will cut into the WebRTC Neteq theme, Neteq is one of the two core technologies of audio technology in WEBRTC (another core technology is the front and back processing of audio, including AEC, ANS, AGC etc, commonly known as 3 a algorithm).
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
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
WebRTC represents the best technology in the field of real-time communication since the date of birth. But for a long time, it supported the video encoder only VP8, and later with H265/VP9 as the representative of the birth of the next generation of video Encoders, WebRTC appeared VP9 Codec. The most widely used H264 has been kept out of sight. Until Cisco announces its H264 codec open source for OpenH264,
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. ”
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
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
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
"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 IP in 2010 Solutions company to get a technology, Google Open source of this technology! ”Just for the more than 60 million dollar technology, it is worth studying!
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
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