webrtc to rtmp

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).

Use librtmp for H264 and AAC live(ext: http://www.codeman.net/2014/01/439.html)1. Frame Division 1.1 H. FramesFor H. 00 00 00 01 or 00 00 01 for each frame.For example, the following H264 file fragment this packet three frame dataE0 9F 96 10 00 (C0)In the F1 2A 00 00 00 013C (CE) 5D DC E9 BDE6 D9 B7 2C D8 D9 EE EF ...The first frame is the xx-C0-DA-E0-9F, which was F1 2A.The second frame is xx 3C 80The third frame is the FF FF 5D DC E9 BD E6 D9 B7 2C D8 D9 EE EF.The frame types are:Nal_slice = 1

HLS, then have the SRS to replace Nginx idea. In the construction of Nginx rtmp service is referred to the predecessors left behind the Nginx build support for the HTTP and rtmp protocol, one of the streaming media server, two or three, because it is the first time in the Nginx to add Nginx-rtmp-module module, the feeling and SRS compared with the relative annoy

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

: This article mainly introduces the nginx Reverse proxy TCP to fetch the RTMP stream. if you are interested in the PHP Tutorial, refer to it. Nginx Reverse proxy TCP, get RTMP stream I. descriptionNginx only supports HTTP reverse proxy by default. to support TCP reverse proxy, you must add the tcp proxy module nginx_tcp_proxy_module. II. raw materials1. nginx_tcp_proxy_module:Wget https://github.com/yaowe

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

: This article describes how to build an rtmp VOD system for nginx development (3). For more information about PHP tutorials, see. 1:/create multimedia/rtmp in the/usr/local/nginx/html/Directory 2: copy the file 1. flv to the Directory 3: open usr/local/nginx/conf/nginx. conf and modify it as follows: Rtmp {# Rtmp

Overview: TCP Establish a connection. rtmp shook hands. • The client and server agree on setting up an rtmp connection. • Create RTMP streams • The client and server agree on play or publish. • The client begins transmitting data to the server. • Message structure Brief tcp establish a connection TCP three times handshake rtm