safari webrtc

1, about WEBRTCWebRTC is a very popular project. The first problem encountered is the WEBRTC compilation problem.Fortunately, a company has helped compile and put it in Maven's repo.Address:Http://mvnrepository.com/artifact/io.pristine/libjingleThe update is very fast, and WEBRTC the official Basic sync update.2,android DemoThe project is also within the pristine project:Https://github.com/pristineio/apprtc

The combination of gstreamer and webrtc is a little breakthrough, gstreamerwebrtc Today, I found a fork killer in gstreamer, and quickly came up with a general framework and solution plan, using the gst-inspector to perform object introspection attribute detection first, then, the gst-launcher tool is used for Pipeline Test. Finally, the channel Logic Source Code is implemented using c to implement webrtc-

from a downhill racing race. Most of the video remains the same, except that the moving parts, i.e. the car and the audience, need to be encoded as P-frames without changing the video. The I frame is generated as a new reference point for P frames. Usually create an I-frame when the image changes very much, such as: panning, scene switching, a large number of actions, sudden disappearance and other scenes. error recovery mechanism:it is suitable for the error recovery mechanism of various packe

The previous article (WEBRTC Audio-related Neteq (a)) is an overview of Neteq, know that it is mainly used to solve the network delay jitter drops and other problems to improve the voice quality, but also know that it has two large units of MCU and DSP components. MCU is mainly received from the network of voice RTP packets into the packet buffer, but also based on the calculated network delay and jitter buffer delay and the feedback from the DSP unit

WebRTC (Web Real time communication) is not Google's original technology, in 2010, Google bought about $68.2 million for VoIP softwareDeveloper Global IP Solutions Company, open source WEBRTC real-time communication project.Voice engine is the gips of voice communication, it is mainly through a series of transmission control to achieve low bandwidth transmission of real-time voice, Gips speech engine hasa w

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-

WEBRTC Voice Overall framework Figure One voice overall frame diagram As shown above, the entire processing frame of the audio is responsible for the transmission of the peer data in addition to the Ligjingle, mainly the Voe (Voice Engine) and the channel adaptation layer Figure II Creating a data communication channel timing diagramThe image above is the local sideComplete process, Voe is created by Createmediaengine_w, the channel adaptation layer

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC provides real-time, web-based audio and video data interoperability, but WEBRTC can also run as a native app on a mobile platform, WEBRTC is a set of media frameworks, implemented in C + +, and officially ported to mobile platforms, including Android,ios, Platform-corresponding development language can be directly deve

First, an overview of the WEBRTC audio processing flow, see:WEBRTC The audio session is abstracted into a channel channels, such as A and b for audio calls, a needs to establish a channel and b for audio data transmission. There are three channel, each channel contains codec and rtp/rtcp send function.In the case of a channel, the application will contain three active threads, a recording thread, an audio receive thread, and a playback thread.1) Recor

WEBRTC IntroductionWebRTC (Web real-time Communications) is a protocol that allows us to implement peer-to-peer on the browser. We can use this protocol to transfer text, voice, video and file content. This article has recorded some personal understanding of my learning process. It is highly recommended to read the documentation for MDN for systematic learning.Simple processFirst, we have a bit a and point B want to communicate with each other. At the

WEBRTC's echo Cancellation algorithm (AEC,AECM) has several important modules:1. Echo Delay estimation2.NLMS3.NLP4.CNG5. Double-ended detection (DT)The following are respectively described:(1) Echo delay estimationecho Delay Length: Based on correlated time delay estimation algorithm (including: Based on the speech signal autocorrelation pitch period): Echo cancellation site, time delay search range is large.WEBRTC's echo delay estimation, which is based on the algorithm of Gips chief scientist

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

The most recent time you've been doing WEBRTC-based Android apps has encountered some problems releasing resources, which are now recorded for memos.The official Apprtcdemo is too simplistic and many questions are not involved.1. Releasing the peerconnection resource problem.Scenario: A and B make a call (Video call)Now stop the call in B.Error: After B terminates the call, the terminal a program will exit unexpectedly.Analysis: When A and b make a ca

This article is mainly their own previous research WEBRTC code structure when some information (including ANDROID,IOS,PC), the article from the blog Garden Rtc.blacker, reproduced please explain the source.1, WEBRTC module: Audio data acquisition, sending, receiving, playback call process:2, WEBRTC module: Video data acquisition, sending, receiving, playback call

. RecordeR.connect (context.destination); }3. Real-time data exchangeWEBRTC's other two api,rtcpeerconnection are used for point-to-point connections between browsers, Rtcdatachannel for Point-to-point data communication.The rtcpeerconnection has a browser prefix and is mozrtcpeerconnection in the Chrome browser for the Webkitrtcpeerconnection,firefox browser. Google maintains a library of adapter.js that is used to pump out differences between browsers.var datachanneloptions = { Ordered:false,

We have introduced WebRTC and read the example of https://apprtc.appspot.com/on the Internet (which may need to be accessed through a wall). This example is an application deployed on the Google App Engine and relies on the Gae environment, the background language is Python and also relies on the Google App Engine channel API. Therefore, it cannot be run locally or be expanded. After studying the Python source code in the example, I decided to impleme

As early as 2014 through the WebRTC realized the PC client real-time video voice, then the establishment of peer-to WEBRTC with the Libjingle library, using the Peerconnection API implementation. Later in the Remote Desktop, file transfer requires point-to-point connection, the Libjingle library for a period of time, found a few problems:The 1.libjingle library is built using the XMPP protocol, but our clie