webrtc java

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AEC algorithm in WEBRTC 2

output signal of the filter and the desired response, which is to ask for a gradient. The AEC algorithm in WEBRTC belongs to the piecewise fast frequency domain Adaptive filtering algorithm, partioned block Frequeney Domain Adaptive filter (PBFDAF).To determine whether the distal and proximal end of the conversation, also known as double-ended detection, the following four conditions need to be monitored:1. Only the far end to speak, at this time the

Build Webrtc/licode Server on Mac/ubuntu

gcc-c++ gcc-g77 Flex Bison autoconf automake bzip2-devel zlib-devel ncurses-devel libjpeg-devel Libpng-dev El libtiff-devel freetype-devel pam-devel openssl-devel libxml2-devel gettext-devel pcre-devel3. Installation dependencies3.1 mac./scripts/installmacdeps.sh3.2 Ubuntu./licode/scripts/installubuntudeps.sh4. Installing Licode./scripts/installerizo.sh. /scripts/installnuve.sh5. Mounting Base Example./scripts/installbasicexample.sh6. Run Licode and examples, run at two terminals, or run in the

Android IOS WebRTC Audio Video Development summary (two or three)

mobile video (browse mode)4.1. Environmental requirements:4.1.1. Prepare two Android phones for 4.0 or more. Chrome browser is installed separately4.2. Demonstration steps:4.2.1. All modes of operation are the same as "Demo PC and PC video".five. Demo phone and PC video5.1. Environmental requirements:5.1.1.1 more than 4.0 Android phones.5.1.2.1 computers with a camera and microphone. And the latest version of Chrome is installed .5.2. Demonstration steps:5.2.1. Phone install and open HuRTC4.0,

The key zone of webrtc is the use of the lock.

Webrtc packages criticalsection, which can be used in windows and posix platforms. The basic structure is as follows: In the factory method, you are responsible for the creation of specific class objects, which can be called a simple factory model. A factory is responsible for the creation of all products, different products are created by inputting necessary parameters to the factory. Generally, the created products are related and inherited from an

[WEBRTC] Forcing the use of TCP transport

Previous notes, finishingWEBRTC uses UDP transport by default, but it can also be transmitted over TCP.With TCP transport, servers such as Turnserver,licode,janus and servers are required.1. If you use Turnserver, you only need the client to keep the relaytcp type of candidate, the others are discarded.2. If you are using a server such as Licode,janus, TCP is not supported by default.Because they are used at the bottom of the Libnice open-source Ice library, Libnice supports TCP in newer version

WEBRTC Source Fragment Analysis (1) Audio buffer copy

SOURCE Locationwebrtc/webrtc/modules/audio_device/ios/audio_device_ios.ccFunctionOsstatusAudiodeviceiphone::recordprocessimpl (Audiounitrenderactionflags *ioactionflags,Const Audiotimestamp *intimestamp,uint32_t Inbusnumber,uint32_t innumberframes){...........while (Bufpos {if ((_recordinglength[bufpos] > 0) (_recordinglength[bufpos] {Found the partially full bufferInsertpos = static_castDon ' t need to search more, quit loopBufpos = n_rec_buffers;}e

DirectShow interface in WebRTC Audio/video Module learning

) Minframeinterval The minimum frame duration, in 100-nanosecond units. This value is applies only to capture filters. Maxframeinterval The maximum frame duration, in 100-nanosecond units. This value is applies only to capture filters. Minbitspersecond Minimum Data Rate this pin can produce. Note Deprecated. Maxbitspersecond

Long-polling, Websockets, SSE (server-sent Event), the difference between WebRTC and use

1, first look at the simplest SSE:Only use the SSE-enabled browser (most), the browser built-in EventSource object, the object by default three seconds to refresh the response data.HTML code (taken from W3cschool):DOCTYPE HTML>HTML>Head>Metahttp-equiv= "Content-type"content= "text/html; charset=utf-8" />Head>Body>H1>Get server-side update dataH1>DivID= "Result">Div>Script>if(typeof(EventSource)!=="undefined") {varSource=NewEventSource ("Socket");//parameter for request link Source.onmessage=fun

WEBRTC code for the daytime (eight): Code folder structure

/video_coding//Video Codec processing code, I420, VP8, VP9││├──./modules/video_coding/codecs││├──./modules/video_coding/main//videocodingmodule Processing Code│├──./modules/video_processing//Video processing before and after, Brighten,color enhancement,deflickering. Spatial Resampler, etc.││└──./modules/video_processing/main//videoprocessingmodule│└──./modules/video_render//Video rendering code. Android,ios, Linux, Mac, Windows, Opengles├──./p2p//nat Traversal code. Turn/stun, server and client│

WEBRTC Learning Four: the simplest voice chat

I. Environment Refer to the previous article: WEBRTC Learning Three: recording and playback Two. Implement The network communication protocol is not explicitly specified in the Voiceengine, so voice chat is not possible only by calling the Voiceengine API. Voenetwork provides method registerexternaltransp

WEBRTC Audio-related Neteq (iii): Access packet and delay calculation

In the previous article (WEBRTC Audio-related Neteq (ii): Data structure) Neteq the main data structures, to understand the mechanism of Neteq lay a good foundation. This article is mainly about how the RTP packets received from the network in the MCU are put into packet buffer and taken out from packet buffer, as well as the calculation of the network delay value (optbuflevel) and the jitter buffer delay value (bufflevelfilt). Let's see how RTP voice

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 java_home=/usr/lib/jvm/jdk1.7.0_45Export JRE_HOME=${JAVA_HOME}/JREExport Classpath=.:${java_home}/lib:${jre_home}/libExport Path=${java_home}/bin: $PATH5. Configure the default JDK versionsudo update-alternatives--install/usr/bin/

Java notes Java tutorial translation preface Java introduction Java Native type Java operators summary Java class Java object Java this use Java class members access control Java method return value Java

Java tutorial translation Sequence Java Introduction Build a JSE development environment-install JDK and eclipse Language basics Java Hello World Program Analysis Variable Java Variables Java Native type Conversion of Java

WEBRTC Code Daytime (11): video_coding Module Analysis

1. The main interface provided externallyVideocodingmoduleimpl::incomingpacket, packet processing interface, called Videocodingmoduleimpl after the RTP parsing process::D ecode, processing the decoded interface Vcmreceivecallba After the completion

Analysis of WEBRTC source code architecture

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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 Echo Elimination II

Current results: Speaker read local data, Mic recorded real-time data, can be 90% to eliminate the echo, Aecdelay also need to tune, NS and VAD are normally available. First ensure the AEC normal, next speaker real-time network data, from double

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

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