webrtc audio

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[IOS] Play and record audio using audio Queue Services

To play and record audio on iOS devices, Apple recommends that we use the Avaudioplayer and Avaudiorecorder classes in the Avfoundation framework. Although the usage is simpler, it does not support streaming; This means that before playing the audio, you must wait until the entire audio load is complete before you can start playing the

Audio Unit Component service Reference (Audio Unit Component Services Reference)

This article mainly introduces the composition of AudioUnit This article is made up of your own understanding. If you have any errors, please contact our netizens for correction. Understand the architecture of Audio Unit Before we started, we realized audioUnit through an AudioUnit object, which is an Effect type AudioUnit of this type. This unit consists of many small scopes, scope has element, elementt has channel, stream format, and properties. Do

The method of video and audio time stamping and its audio-video synchronization (playback) principle

how to make video and audio time-stampinghttp://blog.csdn.net/wfqxx/article/details/5497138 1. Video time stamp PTS = inc++ * (1000/fps); Where Inc is a static, initial value of 0, each time the timestamp Inc plus 1. In FFmpeg, the code in pkt.pts= m_nvideotimestamp++ * (M_vctx->time_base.num * 1000/m_vctx->time_base.den); 2. Audio time stamp PTS = inc++ * (frame_size * 1000/sample_rate) The code in FFmpeg

Opening the door to HD: audio of the new generation of audio and video technology

Author: little monk afa published in: Period 2008-03 of Microcomputer Preface "HD" is a complete system, not a single standard or product. In the HD era, both video and audio must meet the HD standard. What is high-definition audio? What are the benefits of high-definition audio compared with normal audio? Starting fro

Using FEC to improve UDP (RTP) audio and video transmission effects _ audio and video codec

Real-time audio and video domain UDP is the king In the Internet, audio and video real-time interaction using the Transport Layer Scheme has TCP (such as: RTMP) and UDP (such as: RTP) two kinds. The TCP protocol can provide a relatively reliable guarantee for data transmission between two endpoints, which is achieved through a handshake mechanism. When the data is passed to the receiver, the receiver checks

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

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

General Audio Protocol Introduction && knowledge and technical parameters of audio coding

(reprint) Common Audio Protocol IntroductionIntroduction and comparison of common audio protocols for conference TV white PaperFirst, the digital audio principle: The sound is actually a kind of energy wave, therefore also has the frequency and the amplitude characteristic, the frequency corresponds to the time axis, the amplitude corresponds to the level axis. U

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

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 subject to change in future versions of browsers a s the spec changes. MediaStream The interface represents a stream of media content. A Stream consists of several tracks, like video or audio tracks.Attri

Audio sampling and audio sampling frequency and Bit Rate in MP3

MP3 is only a type of audio format. However, audio has several important parameters, such as kHz, bit, audio channel, and kbps. the formats are different, and the algorithms are different. Therefore, when the above parameters are the same, the quality of different formats varies greatly. VBR is a kind of dynamic sampling. For more information, see the following d

Audio Enhancement Software SRS Audio Sandbox

Recently my wife gave me a death order, let me the computer's "broken speakers" to replace, she really can not endure to listen to music without surround sound, watching movies without overweight bass, play no game stereo sound, singing KTV simply do not sound ... Vulgar speech: "Head can be broken, blood flow, wife orders can not be lost." "Yes, this time must be excellent to complete the task of wife confessed!" Can change the speaker to spend money, if not change the speaker can also bring th

HTML5 Video player-videojs+audio tag for video, audio and subtitle sync playback

article shows that in the case does not support HTML5, will be flash play, but in support of HTML5 Firefox play MP4, but encountered great difficulty, although the call to Flash, but has been unable to play (but I have always suspected that my Firefox flash has a problem , don't know if it's true). But if you follow Videojs's advice and put in two-format videos, you won't have this problem.In addition to the writing of video also has a special for flash writing, of course, you can also use this

Android Audio Development (2): How to capture a frame of audio

This article focuses on how to capture a single frame of audio data on the Android platform. Before reading this article, it is recommended to read my previous article, "Android Audio Development (1): Fundamentals", because in the audio development process, often involves these basic knowledge, after mastering these important concepts, the development process of

Use audio queue services to play and record audio

To play and record audio on iOS devices, we recommend that you use the avaudioplayer and avaudiorecorder classes in the avfoundation framework. Although its usage is relatively simple, streaming is not supported. This means that before playing the audio, you must wait until the whole audio is loaded to start playing the audio

HTML5 audio function, html5 audio

HTML5 audio function, html5 audioHTML5 audio Functions 1. Web audio Until now, there is still no standard for playing audio on the webpage. Today, most audios are played using plug-ins (such as Flash. However, not all browsers have the same plug-ins. HTML5 specifies a standard method to include

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 talk there is a long way, refueling. Effect Diagram: Real-time audio echo cancellation Flowchart: key points of knowledge: 1, the purpose of ECHO cancellation is to achieve two-way intercom (double ta

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 processing Rtpdata2. Main processing class Modulertprtcpimpl, control module, is a module, oneself can independently handle RTPPACKETIZER/RTPPACKETIZERH264/RTPPACKETIZERVP8 specific format of Zhei processing class rtpdepacketizer/rtpdepacketizerh264/rtpdepa

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 Modulertprtcpimpl::sendoutgoingdataCallback interface after packet processing Rtpdata2. Main processing ClassModulertprtcpimpl, control module, is a modules, you can independently handleRTPPACKETIZER/RTPPACKETIZERH264/RTPPACKETIZERVP8 Zhei processing class for specific formatsrtpde

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