webrtc android

WEBRTC 's echo cancellation (AEC, AECM) algorithm mainly includes 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 include double-ended detection (DT). Considering that the NLMs, NLP and CNG used by WEBRTC belong to the classical algorithm

Recently read a foreigner wrote on the webrtchacks, the main introduction of WEBRTC and WhatsApp transmission mechanism, fine, coupled with their own understanding to summarize,Hope to help everyone, reprint please explain the source, the original

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

In "Let WEBRTC support H264 codec" I provide a priority to use the H264 codec thinking. We can verify it on the browser side. There are three ways to verify: In JS print SDP view Chrome's log chrome_debug.log (see Open Chrome Log) Grab bag using webrtc-internals The first three kinds are no longer introduced, we look at the webrtc-internals. The

Transferred from: http://www.cnblogs.com/fangkm/p/4364553.htmlWEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether usi

WEBRTC is one of the important features of HTML5 support, it is no longer necessary to use audio and video-related clients, directly through the browser's web page to achieve audio and video chat function. And the WEBRTC project is open source, we can use the WEBRTC source code to quickly build their own audio and video chat function. Whether using the

 Harnessing Open Source Library WebRTC Fourth chapter-Compiling Macios edition Author: Adam Acknowledgements: Lao Zhang Date: 2015-4-6 Version: 1.0.0 Welcome reprint, has the question feedback q:2780113541, as far as possible consummates series of tutorials. Update Address: Https://github.com/wpc320/webrtc_doc.git Depot_tools proxy settings Reference old Zhang "the best wall in history download WEBRTC co

2016-11-25 DORAWEBRTC series Wind net WEBRTC series Wind Net Source: Wind NET Series Author: Weizhenwei, fan network columnist Audio and video synchronization is the most intuitive user experience for multimedia products, and is the most basic quality guarantee for audio and video media data transmission and rendering playback. If the audio and video is not synchronized, it is possible to cause delay, lag, etc. that affect the user experience very mu

Wiener filter.////@param aecm [i/n] Handle of the AECM instance.@param supgain [out] (Return value) suppression gain with which to scale the noiseLevel (Q14).////int16_t Webrtcaecm_calcsuppressiongain (aecmcore* const AECM) { In this, you can make a DTD judgment. This is based on the estimated echo signal and the actual input of the echo signal to determine whether the DTD is not. Then is the Wiener filter and the Henning window, as well as the comfort noise generation, does not understand. Dis

ObjectiveThe 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, you'd better evaluate your patience 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 do no

WEBRTC in the Chrome browser demo Many examples, WebRTC source, but in the Firefox browser, the example can not be used, the information on the web said to set the media.peerconnection.enabled to True, However, in the Firefox browser, the default value is True, using the WEBRTC example in Firefox or can not capture local video. Here are the examples found online:

A recent study on how WebRTC speech runs on iOS found that the voice_engine of WebRTC has implemented iOS-related classes, but encountered a series of problems in specific applications. After several days of hard work, finally, we solved a series of problems and successfully realized recording and playing local loop in the simulator. Compile the testProgramIn the process, we plan to use the libjingle Libr

I have only recently started to study webrtc deeply. If I have any questions, please leave a message.How to generate WEBRTC vs engineering under Windows see my last article.But when I modify the project, such as adding cc and H files, adding a third-party dependency library, you will find that the VS modification is useless. VS can only be used when a code reader and editor are available.This time we need t

Turn from: http://blog.csdn.net/lixiaowei16/article/details/53407010 Audio and video synchronization is related to the most intuitive user experience of multimedia products, audio and video media data transmission and rendering playback of the most basic quality assurance. If the audio and video is not synchronized, it may cause delays, such as cotton, etc. very affect the user experience phenomenon. Therefore, it is very important. Generally speaking, the audio and video synchronization maint

The bandwidth assessment (BWE) is perhaps the most critical module in the WEBRTC video engine, which determines the amount of video data that can be generated when network congestion is not raised in video traffic. Early bandwidth assessment algorithms are relatively primitive, mostly based on packet loss estimation, the basic strategy is to gradually increase the amount of data sent, until the loss of packets detected. In order for the sender to lear

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

Google's first integration of WebRTC in the Chrome Dev release released this January was a source of widespread concern. Today, Google published a roadmap for the development of WebRTC technology in its blog.WebRTC is a technology for real-time video and audio communication inside the browser, and Google acquired a technology in 2010 to acquire Global IP Solutions. The technology is based on the WHATWG prot