webrtc server

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

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

Recently in-house training for companies, mainly on instant messaging and mobile video calls, including Android and Android,ios with Ios,android and iOS, as well as mobile and PCDo not fight unprepared for the war, so carefully collated a more detailed outline, the following outline for mobile video calls,Although the content of the talk is not directly shared, but this outline for everyone to clear the idea of the video call will be helpful,Of course, if you think this outline is missing or fee

Resources:Http://bucephalus.org/text/CanvasHandbook/CanvasHandbook.html#getcontext2dHttps://developer.mozilla.org/zh-CN/docs/Web/HTML/CanvasHttp://www.w3school.com.cn/html5/html5_canvas.aspHttps://developer.mozilla.org/zh-CN/docs/Web/API/HTMLCanvasElementis a new element of HTML5, you can use JavaScript scripts to draw graphics. For example: paint, synthesize photos, create animations and even real-time video processing and rendering.Mozilla programs are supported from Gecko 1.8 (Firefox 1.5) .

The link address of the original English text is: Https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/OverviewWEBRTC is a technology that collaborates with a number of associated APIs and protocols to support the exchange of data and media information between two or more terminals. This article provides an introduction to these APIs and provides functionality.RtcpeerconnectionYou need to connect the two terminals before the media can be exchanged or the data channel is set up. The comple

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version number TELEMCU added Android phone-side WEBRTC video conferencing capabilities, Android phone installed Chrome/firefox browser after loading TELEMCU Webrtcclientteleweb can directly participate in video conferencing,At the same time, TeleWeb can support up to two webrtcclient-to-peer communication, such as the following:TeleWeb Test Address:Http://ope

anyway.-Not really, the project over there is not finished yet ...Sad and hurried, so:As party A: the choice is most important to the person. When you do not know whether or not to choose the right person, you can only start small projects test water or have a backup plan. In addition to the actual cost, once you find the right person, do not be too stingy and timely payment, so as to have a winning result.As party B: must be delivered on schedule, must be honest. If you can't deliver it on sch

services; loss of use, DATA, or profits; * or business interruption) HOWEVER CAUSED AND ON ANY TH Eory of liability, * whether in contract, strict liability, or tort (including negligence or * OTHERWISE) arising in any way out of the use of this software, even if * advised of the possibility of such damage. */# ifndef TALK_BASE_BASICTYPES_H _ # define TALK_BASE_BASICTYPES_H _ # include The above Code defines the basic types, as well as the hardware architecture, in byte order. It will be use

WebRTC Code read (10): rtp_rtcp module analysis, webrtcrtp_rtcp1. Call interface RtpReceiverImpl: IncomingRtpPacket call interface ModuleRtpRtcpImpl: RtpData2. the main processing class ModuleRtpRtcpImpl, control Module, is a Module, you can independently process RtpPacketizer/RtpPacketizerH264/handler specific Format Decoding handler class RtpDepacketizer/Resolver/handler/specific format parsing RTP Header Processing class RtpReceiverImpl accept RTP

higher quality network video at limited bandwidth. For most professionals, the h.265 coding standard is not unfamiliar, it is itu-tvceg after the development of the video coding standards. The h.265 standard mainly revolves around the existing video coding standard, which, in addition to preserving some of the original technologies, increases the correlation between the code stream, the encoding quality, the delay, and the complexity of the algorithm. The main contents of h.265 research include

The Audio_device is a WEBRTC audio device module. Encapsulates audio device-related code for each platform Audio device encapsulates two sets of sound code in Android. 1. Use JNI to invoke Java's media. 2. Operate directly through the native C interface of the OpenSL es. The native interface is naturally more efficient, but the downside is that OpenSL requires Android 2.3+. OpenSL ES (Open sound Library for Embedded systems) is a hardware audio accel

When using WEBRTC on the Android layer, the UI changes are triggered by the native layer callback, such as when to draw the other's video window, when to indicate that both connections have been established, etc...I'm going to list what I know now for the memo.Onaddstream (), which indicates that the associated media stream has been initialized successfully (but does not establish a connection), usually at this time display the other side of the video

, Videocodec); ASSERT (IRet==ret_success); IRet= m_viecapture->Connectcapturedevice (Icaptureid, m_channelid); ASSERT (IRet==ret_success); IRet= m_viertp_rtcp->setrtcpstatus (M_channelid, webrtc::krtcpcompound_rfc4585); ASSERT (IRet==ret_success); IRet= m_viertp_rtcp->Setkeyframerequestmethod (M_channelid, webrtc::kviekeyframerequestplirtcp); ASSERT (IRet==ret_success); IRet= M_viertp_rtcp->settmmbrstatus (

The purpose of this article is to save you 10 hours (or more) of your life, or to waste 10 minutes. WEBRTC's compilation has been called a nightmare as a large cross-platform base library that Google has frequently updated. If you happen to want to compile WEBRTC under Windows, then you'd better evaluate your patience and IQ in advance. As of now, I have tried almost all the articles in the Chinese blog community, which can be said to have failed. I d

This article mainly introduces FFmpeg, the article comes from the blog Garden Rtc.blacker, supports the original, the reprint must explain the source, the individual public number blacker, more see Www.rtc.helpDescriptionPS1: If you start learning audio and video directly from WEBRTC, you may not have heard of ffmpeg, and you don't need it, but as you improve your personal abilities, you'll find it really useful.As far as I am currently exposed to the

Recently finally updated the PC version of the WEBRTC, summarized under what adjustments, the article from the blog Garden Rtc.blacker, support the original, reproduced please explain the source.Figure 1: Solution Engineering Structure Comparison:Description1, the biggest adjustment is to remove the Videoengine module, the relevant effects are as follows:1.1, Webrtcdemo inside removed video calls, voice calls still exist, but the removal is a matter o

;tm_min, pbacktime->tm_sec); - -Write (WT_FD, flag,1);//Reset Watchdog Feed the dog inAlarm2); - return; to } + - the intMain () * { $ CharFlag ='V';Panax Notoginseng intret; - intTimeout = the; the + if(Sig_err = =signal (SIGALRM, sigalarm)) A { thePerror ("Signal (sigalarm) Error"); + } - $WT_FD = open ("/dev/watchdog", O_RDWR); $ if(WT_FD 0) - { -printf"Fail to open watchdog device!\n"); the } - ElseWuyi

Today let me find a gstreamer of a bull fork of the killer, the mind immediately thought of a general framework and plan, with Gst-inspector first object introspection property detection, and then sacrificed Gst-launcher Broadsword for pipeline test, and finally use C to achieve the pipeline logic source code , you can implement WEBRTC-based video surveillance and live streaming services. Real-time two-person video call or multi-person meeting, after