jitsi webrtc

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WebRTC audio capture, encode, send

Audiodevicelinuxpulse :: recthreadprocessAudiodevicelinuxpulse :: ProcessrecordeddataAudiodevicelinuxpulse::readrecordeddataProvide data to Voiceengineif (Processrecordeddata (_recbuffer, numrecsamples, recdelay) = =-1){We have stopped

Webrtc lock, easy to remember

Today, I want to capture some excellent code, but the dependency between files is too large and difficult to find. It seems that I found a place. Lock usage: 1. multi-threaded access to the same variable requires locking. 2. A lock is a kind of

Webrtc debugging control common. h file

Common. h Mainly defines some compilation Conditions1. Disable the 4355 warning. 2. The definition of sdtmax has no value. 3, the macro of ARRAY_SIZE, evaluate the array size 4, ENABLE_DEBUGUse ASSERT (x) and VERIFY (x) macros when ENABLE_DEBUG

WEBRTC Code Daytime (10): rtp_rtcp module Analysis, webrtcrtp_rtcp

Transferred from: http://www.bkjia.com/Androidjc/1020017.html1. The main process interface provided externallyThe calling interface of the packet Rtpreceiverimpl::incomingrtppacketThe calling interface of the contract

WEBRTC Source Analysis Three: Video processing flow

Transferred from: http://blog.csdn.net/neustar1/article/details/19480863Text describes the process of video processing. The two-way video session video signal flow process is shown in Figure 1.Figure 1 Video FlowTake the video session as an example,

Android IOS WebRTC Audio Video Development summary (27)

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

WEBRTC-based Media Library test code and interface introduction

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,

WEBRTC combined with canvas screenshot

Look directly at the code. CSS base weak chicken, will see it. Learn slowlyDOCTYPE HTML>HTML>Head> MetaCharSet= "Utf-8"> title>Canvastitle> styletype= "Text/css">Body{Display:Block;width:50%;Margin-left:Auto;Margin-right:Auto;

Webrtc–getusermedia & Canvas

The following is a feature implemented using the Getusermedia interface and the canvas's DrawImage method (capturing a frame in the video).The basic idea is this : Getusermedia gets a mediastream, stream Stream as the input source for

WEBRTC Encoder Bandwidth Tuning __JS

uint32_t mediaoptimization::settargetrates ( uint32_t target_bitrate, uint8_t fraction_lost, int64_t Round_trip_time_ms, vcmprotectioncallback* protection_callback) { criticalsectionscoped lock (crit_sect_. Get ()); ...

AGC automatic gain control of WEBRTC

When the loudness of the voice to adjust the need for voice automatic gain (AGC) algorithm processing , voice chat will use this algorithm. The simplest hard gain processing is to multiply all audio samples by a gain factor, which is also

26 Web conferencing/video conferencing open source software

Last updated: Video conferencing software Vmukti 1.1.3.1 Release posted 6 year ago Distance Education Platform BigBlueButton BigBlueButton is an online video conferencing system developed using ActionScript, or a distance education system, with features such as online PPT presentations, video communication and voice communication, as well as text exchange, hands-on presentations, and more, especially for online teaching supports a wide

2015GitWebRTC Compilation Transcript 8

2015.07.20 Common_video compiled by, it has a reference to LIBYUV[1309/1600] CXX OBJ/WEBRTC/COMMON_VIDEO/LIBYUV/COMMON_VIDEO.SCALER.O[1310/1600] CXX OBJ/WEBRTC/COMMON_VIDEO/COMMON_VIDEO.I420_BUFFER_POOL.O[1311/1600] CXX OBJ/WEBRTC/COMMON_VIDEO/COMMON_VIDEO.VIDEO_FRAME.O[1312/1600] Librtc_sound.a[1313/1600] CXX OBJ/WEBRTC

Install JitsiInstantMessenger (Jisti Instant Messenger) in Ubuntu)

Jitsi is an open-source, multi-platform audio/video network telephone and instant messaging software written in Java. It supports some of the most popular instant messaging and telecommunications protocols, such as SIP, Jabber/XMPP (which supports Facebook and GoogleTalk), AIM, ICQ, MSN, Yahoo! Messenger. Jitsi is also considered a good replacement for Skype. Install Ji

Chromium's backtrace record

FFmpeg after processing the video stream, the upper WebRTC call error, you can see the WEBRTC call process:BackTrace:Webrtc::rtpfragmentationheader::copyfrom [0x5813cad2+18] (d:\workspace\chromium_build\src\third_party\webrtc\ modules\interface\module_common_types.h:283)Webrtc::rtppacketizerh264::setpayloaddata [0x5813

2015GitWebRTC Compilation Transcript 3

2015.05.17 Librtprtcp compiled by[702/1600] CXX OBJ/WEBRTC/MODULES/RTP_RTCP/SOURCE/RTP_RTCP.BITRATE.O[703/1600] CXX OBJ/WEBRTC/MODULES/RTP_RTCP/SOURCE/RTP_RTCP.FEC_RECEIVER_IMPL.O[704/1600] CXX OBJ/WEBRTC/MODULES/RTP_RTCP/SOURCE/RTP_RTCP.RECEIVE_STATISTICS_IMPL.O[705/1600] CXX OBJ/WEBRTC/MODULES/RTP_RTCP/SOURCE/RTP_RTC

2015GitWebRTC Compilation Transcript 6

2015.07.20 Libbitrate_controller compiled byDepending on system_wrappers Lib, you need to be aware of writing test code.[425/1600] CXX OBJ/WEBRTC/MODULES/BITRATE_CONTROLLER/BITRATE_CONTROLLER.BITRATE_CONTROLLER_IMPL.O[426/1600] CXX OBJ/WEBRTC/MODULES/BITRATE_CONTROLLER/BITRATE_CONTROLLER.BITRATE_ALLOCATOR.O[427/1600] CXX OBJ/WEBRTC/MODULES/BITRATE_CONTROLLER/BITR

Do mobile video Call software, roughly see the existing open source software

embedded and desktop systems. The framework is written using ansci-c and is very portable. and has been designed to be very lightweight and effective in embedded systems with low memory and low processing power. The Idoubs feature on the Apple system is based on this framework. Most of the audio and video encoding formats are supported (H264 (VIDEO), VP8 (video), ILBC (audio), pcma,pcmu,g722,g729). NAT supports Ice (Stun+turn)2) Effect measurementTest environment: Two machines in the company's

Ubuntu14.04 compile WebRTCForAndroid code

Recently, a real-time communication project for audio and video chats was developed based on Google's open-source WebRTC project. Some problems were encountered during the download of WebRTC code, which was recorded here, we also hope to help the children's shoes who encounter similar problems while downloading and compiling WebRTC code. 1). Prepare the environme

Android's Open source phone/Communication/im Chat project complete

One, the Android XMPP client BeemBeem is an XMPP (jabber) client software that runs on the Android mobile platform and is compatible with standard XMPP servers and services such as Ejabberd, OpenFire, Facebook, and Gmail. Supports SASL, proxies (SOCKS4, SOCKS5, HTTP), DNS SRV, and chatting with a specific resource.Project home: http://beem-project.com/Second, Android chat program jchat4androidAndroid Jchat is a location-based chat software on an Android phone, using a peer-messaging mechanism.

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