webrtc broadcast

WEBRTC IntroductionWEBRTC provides three types of APIs: MediaStream, namely Getusermedia Rtcpeerconnection Rtcdatachannel Getusermedia has been supported by Chrome, opera and Firefox.rtcpeerconnection is currently supported by Chrome, opera and Firefox. Chrome and opera provide an interface named Webkitrtcpeerconnection,firefox with the name Mozrtcpeerconnection.Rtcdatachannel is only available in Chrome, Opera 18 and Firefox 22

WEBRTC is a set of new standards for media data transmission based on the browser side, introducing a number of new concepts, including Dtls, SDEs, DTLS-SRT, ice, turn, Rtp-mux, BWE, FEC jsep, Tricle-ice and other terms,This article first said Dtls, DTLS-SRTPDTLS: Full name Datagram Transport Layer Security, which is UDP + secure, the datagram layer is safe, DTLS employs TLS security mechanism, but is more lightweight,

WEBRTC Technology Group: 234795279 1. Voiceengine CODEC data structure WEBRTC, a struct struct codecinst is used to represent a specific audio codec object: struct Codecinst { int pltype; Payload Type Payload char plname[32];//payload name payload, 32 characters representing int plfreq; Payload frequence Load Frequency int pacsize; Packet size package int channels; Chan

WebRTC FEC (forward error correcting code) is an important part of its QoS, which is used to recover original packets when network drops, reduce retransmission times, reduce delay and improve video quality. It is an implementation of the RFC 5109 standard. Below, we will delve into its principles. redundant Coding To understand the FEC in WEBRTC, you first need to understand the red Packet. The so-called Re

Send and receive broadcasts across applications. In the same case as in the same application, you only need to add a permission and configure the Android: process attribute of the receiver. In the broadcast sending application: Java code Intent intent = new intent ("info. zhegui. Er er. interprocess "); Sendbroadcast (intent ); Note that you must add the broadcast receiving permission in manifest. xml

[Project 1 live broadcast]☞1. Create a project and configuration, and broadcast a project1. Create a project directory Specify a directory when creating a project. Specify the directory to define your own architecture. The structure is well-defined to facilitate searching. Create the Classes directory (which contains the custom Classes) and continue to create the directory: Show: Display Module, mainly re

Cool Dog stars live tutorial:1. This live video requires a camera, so we need to connect the camera to the computer and debug it;2. Register stars live broadcast users;Http://fanxing.kugou.com /,3. After entering the room, click [I want to start broadcasting] to quickly access our own live broadcasting room;4. In the live broadcasting room, select a video device;5. After selecting the appropriate poster, we can start shooting the Live Poster. Then, we

Preparations for live mobile games:We must connect the iphone and PC to the same Wi-Fi network. We can use Airplay of ios.Install the itools tool on your computerHow to broadcast a mobile game through dub Li's livestream Ji?1. Now, in the itools toolbox-device management, search "Apple screen recorder" to find and install and use the tool.2. In the mobile phone, find the iphone status bar and click Airplay. Then, select the corresponding

How to play the Live VR live pepper 1, through the Advisory Pepper Live customer service, found that if you want to open VR live, the need for host computer to open live. VR Webcam link computer. And fans need to wear VR glasses. 2, in addition, if the audience want to watch the stereo live, can not directly find the VR classification of live, because there is no such live room, so want to experience the small partners need to wait Chinese prickly ash live VR stereo Live play detailed

mobile video (browse mode)4.1. Environmental requirements:4.1.1. Prepare two Android phones for 4.0 or more. Chrome browser is installed separately4.2. Demonstration steps:4.2.1. All modes of operation are the same as "Demo PC and PC video".five. Demo phone and PC video5.1. Environmental requirements:5.1.1.1 more than 4.0 Android phones.5.1.2.1 computers with a camera and microphone. And the latest version of Chrome is installed .5.2. Demonstration steps:5.2.1. Phone install and open HuRTC4.0,

Webrtc packages criticalsection, which can be used in windows and posix platforms. The basic structure is as follows: In the factory method, you are responsible for the creation of specific class objects, which can be called a simple factory model. A factory is responsible for the creation of all products, different products are created by inputting necessary parameters to the factory. Generally, the created products are related and inherited from an

Previous notes, finishingWEBRTC uses UDP transport by default, but it can also be transmitted over TCP.With TCP transport, servers such as Turnserver,licode,janus and servers are required.1. If you use Turnserver, you only need the client to keep the relaytcp type of candidate, the others are discarded.2. If you are using a server such as Licode,janus, TCP is not supported by default.Because they are used at the bottom of the Libnice open-source Ice library, Libnice supports TCP in newer version

SOURCE Locationwebrtc/webrtc/modules/audio_device/ios/audio_device_ios.ccFunctionOsstatusAudiodeviceiphone::recordprocessimpl (Audiounitrenderactionflags *ioactionflags,Const Audiotimestamp *intimestamp,uint32_t Inbusnumber,uint32_t innumberframes){...........while (Bufpos {if ((_recordinglength[bufpos] > 0) (_recordinglength[bufpos] {Found the partially full bufferInsertpos = static_castDon ' t need to search more, quit loopBufpos = n_rec_buffers;}e

) Minframeinterval The minimum frame duration, in 100-nanosecond units. This value is applies only to capture filters. Maxframeinterval The maximum frame duration, in 100-nanosecond units. This value is applies only to capture filters. Minbitspersecond Minimum Data Rate this pin can produce. Note Deprecated. Maxbitspersecond

1, first look at the simplest SSE:Only use the SSE-enabled browser (most), the browser built-in EventSource object, the object by default three seconds to refresh the response data.HTML code (taken from W3cschool):DOCTYPE HTML>HTML>Head>Metahttp-equiv= "Content-type"content= "text/html; charset=utf-8" />Head>Body>H1>Get server-side update dataH1>DivID= "Result">Div>Script>if(typeof(EventSource)!=="undefined") {varSource=NewEventSource ("Socket");//parameter for request link Source.onmessage=fun

/video_coding//Video Codec processing code, I420, VP8, VP9││├──./modules/video_coding/codecs││├──./modules/video_coding/main//videocodingmodule Processing Code│├──./modules/video_processing//Video processing before and after, Brighten,color enhancement,deflickering. Spatial Resampler, etc.││└──./modules/video_processing/main//videoprocessingmodule│└──./modules/video_render//Video rendering code. Android,ios, Linux, Mac, Windows, Opengles├──./p2p//nat Traversal code. Turn/stun, server and client│

I. Environment Refer to the previous article: WEBRTC Learning Three: recording and playback Two. Implement The network communication protocol is not explicitly specified in the Voiceengine, so voice chat is not possible only by calling the Voiceengine API. Voenetwork provides method registerexternaltransp

In the previous article (WEBRTC Audio-related Neteq (ii): Data structure) Neteq the main data structures, to understand the mechanism of Neteq lay a good foundation. This article is mainly about how the RTP packets received from the network in the MCU are put into packet buffer and taken out from packet buffer, as well as the calculation of the network delay value (optbuflevel) and the jitter buffer delay value (bufflevelfilt). Let's see how RTP voice

There are two types of echoes in voice calls:1. Circuit echo (already resolved)2. Acoustic echoTwo echo cancellation modules are designed in the WEBRTC source code:1.AEC (Acoustic Echo canceller): PC side2.AECM (Acoustic Echo Canceller mobile): MobileAECM:Causes of acoustic Echo:The voice of the proximal speaker is picked up by his microphone and transmitted to the far end via the network,The sound from the remote speaker is picked up by the microphon

() {var newindex = (Filters.indexof (canvas.classname) + 1)% Filters.length; Canvas.classname = Filters[newindex];} Navigator.getusermedia = Navigator.getusermedia | | Navigator.webkitgetusermedia | | navigator.mozgetusermedia;//WebRTC Constraintsvar constraints = {audio:false, video:true};var video = Document.queryse Lector ("video");//MediaStream as Video input function Successcallback (stream) {window.stream = stream;//Stream available to console