webrtc android

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

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

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

 Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4401075.html The first two blog posts complete the WEBRTC audio and video collection module, and the next step is to introduce the key audio and video coding modules. However, before introducing the audio and video coding module, we need to introduce the channel concept, and the transmission flow of each WEBRTC data is encapsulated into a c

gcc-c++ gcc-g77 Flex Bison autoconf automake bzip2-devel zlib-devel ncurses-devel libjpeg-devel Libpng-dev El libtiff-devel freetype-devel pam-devel openssl-devel libxml2-devel gettext-devel pcre-devel3. Installation dependencies3.1 mac./scripts/installmacdeps.sh3.2 Ubuntu./licode/scripts/installubuntudeps.sh4. Installing Licode./scripts/installerizo.sh. /scripts/installnuve.sh5. Mounting Base Example./scripts/installbasicexample.sh6. Run Licode and examples, run at two terminals, or run in the

After a period of project validation, the first version of the interface can be fixed version. Meet the general requirements of the project is no problem, the interface is very clear, the gaze is also written very clear, we have to take to test it, the Android version number is still in the verification. Suppose there is any problem or need source code. Can leave a message or contact me, qq:3049230939. Interface and test procedure: http://download.csd

Figure Description: Moving, backup after the launch of Android virtual machine appears Solution to the problem of panic:could not open D:\java2\android\android-sdk-windows\.android\avd\test.ini A few days ago, I am sorting some files, the previously installed on the Android