janus webrtc

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WebRTC Configuring the Environment

Copying files to the specified file path Cp-rf/home/leehongee/leehongee/jdk1.7.0_45/usr/lib/jvm Create folder mkdir JVM modifying environment variables sudo gedit/etc/profileAdd to#set Java EnvironmentExport

WEBRTC needs, configure its own Turn/stun service

1, server environment Ubunutu 16.04LTS; 2, the installation needs to rely on sudo apt-get-y install SQLite libsqlite3-dev libevent-dev Libssl-dev 3. Download Turnserver Source code wget http://turnserver.open-sys.org/downloads/v4.5.0.7/turnserver-4.5

WEBRTC Encoder Bandwidth Adjustment __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 ()); ...

WEBRTC Learning--mediastream and Mediastreamtrack

This was an experimental technologyBecause This technology ' s specification have not stabilized, check the compatibility table for the proper prefixes to use I n various browsers. Also Note the syntax and behavior of an experimental technology are

WEBRTC Code Daytime (10): RTP_RTCP Module Analysis

1. The main process interface provided externallyThe calling interface of the receiving packet Rtpreceiverimpl::incomingrtppacket the calling interface of the package Modulertprtcpimpl::sendoutgoingdata the callback interface after the packet

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

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

2018-07-23 Headlines Latest

/IP and HTTP have a deep understanding; 3, familiar with WEBRTC, FFmpeg, Licode, Kurento, Janus, Mediasoup and other audio and video tools; 4, familiar with H264, H265, Opus, VP8 and other codec, familiar with RTP, rtmp, RTSP, SIP and other transmission protocols; 5, the voice of the relevant algorithm optimization experience is preferred, such as NS,VAD,AGC,AEC. Jobs audio and video real-time communication

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

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

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