鍍金池/ 教程/ Android/ WebRTC
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WebRTC

WebRTC

目標(biāo)

  • 便捷的為web應(yīng)用添加視頻聊天、p2p數(shù)據(jù)傳輸功能;
  • API開源、免費(fèi)、標(biāo)準(zhǔn)化,嵌入到瀏覽器中,并且比現(xiàn)有插件技術(shù)更高效;

API

  • MediaStream (aka getUserMedia)
  • RTCPeerConnection
  • RTCDataChannel

  • MediaStream
    具有input和output,input獲取視頻數(shù)據(jù)、音頻數(shù)據(jù);output將數(shù)據(jù)展示到video/audio標(biāo)簽,或發(fā)送到peer;

      var stream = navigator.getUserMedia()
      stream.getAudioTracks()
      stream.getVideoTracks()
    
      URL.createObjectURL() 
  • Constraints
    applyConstraints()函數(shù)為getUserMedia設(shè)置限制條件,支持分辨率、長(zhǎng)寬比、前后攝像頭、幀率、寬高等;
    在一個(gè)標(biāo)簽頁中設(shè)置了Constraints之后,后面打開的標(biāo)簽頁都受影響;
  • Screen and tab capture chrome目前支持屏幕錄制、瀏覽器標(biāo)簽頁錄制API;
  • Signaling: session control, network and media information
    WebRTC使用RTCPeerConnectionAPI進(jìn)行peer間通訊,signaling進(jìn)程負(fù)責(zé)維護(hù)通信、發(fā)送控制信息;
    • Session control messages: to initialize or close communication and report errors.
    • Network configuration: to the outside world, what's my computer's IP address and port?
    • Media capabilities: what codecs and resolutions can be handled by my browser and the browser it wants to communicate with?

WebRTC on Android