示例

[TOC] > [sample](https://webrtc.github.io/samples/) ## 概述 访问媒体设备 * [基本的getUserMedia演示](https://webrtc.github.io/samples/src/content/getusermedia/gum/) * [将getUserMedia与画布一起使用](https://webrtc.github.io/samples/src/content/getusermedia/canvas/) * [将getUserMedia与canvas和CSS过滤器一起使用](https://webrtc.github.io/samples/src/content/getusermedia/filter/) * [选择相机分辨率](https://webrtc.github.io/samples/src/content/getusermedia/resolution/) * [仅音频的getUserMedia（）输出到本地音频元素](https://webrtc.github.io/samples/src/content/getusermedia/audio/) * [仅音频的getUserMedia（）显示音量](https://webrtc.github.io/samples/src/content/getusermedia/volume/) * [记录流](https://webrtc.github.io/samples/src/content/getusermedia/record/) * [使用getDisplayMedia进行屏幕共享](https://webrtc.github.io/samples/src/content/getusermedia/getdisplaymedia/) * [控制相机的平移，倾斜和缩放](https://webrtc.github.io/samples/src/content/getusermedia/pan-tilt-zoom/) ## 设备： 查询媒体设备 * [选择摄像头，麦克风和扬声器](https://webrtc.github.io/samples/src/content/devices/input-output/) * [选择媒体源和音频输出](https://webrtc.github.io/samples/src/content/devices/multi/) ## 流捕获： 从画布或视频元素流 * [从视频元素流到视频元素](https://webrtc.github.io/samples/src/content/capture/video-video/) * [从视频元素流向对等连接](https://webrtc.github.io/samples/src/content/capture/video-pc/) * [从画布元素流到视频元素](https://webrtc.github.io/samples/src/content/capture/canvas-video/) * [从画布元素流向对等连接](https://webrtc.github.io/samples/src/content/capture/canvas-pc/) * [记录来自canvas元素的流](https://webrtc.github.io/samples/src/content/capture/canvas-record/) * [通过内容提示指导视频编码](https://webrtc.github.io/samples/src/content/capture/video-contenthint/) ## [RTCPeerConnection：](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection) 控制对等连接 * [基本对等连接演示](https://webrtc.github.io/samples/src/content/peerconnection/pc1/) * [纯音频对等连接演示](https://webrtc.github.io/samples/src/content/peerconnection/audio/) * [随时更改带宽](https://webrtc.github.io/samples/src/content/peerconnection/bandwidth/) * [通话前更改编解码器](https://webrtc.github.io/samples/src/content/peerconnection/change-codecs/) * [升级通话并打开视频](https://webrtc.github.io/samples/src/content/peerconnection/upgrade/) * [一次多个对等连接](https://webrtc.github.io/samples/src/content/peerconnection/multiple/) * [将一台PC的输出转发到另一台PC](https://webrtc.github.io/samples/src/content/peerconnection/multiple-relay/) * [Munge SDP参数](https://webrtc.github.io/samples/src/content/peerconnection/munge-sdp/) * [建立对等连接时使用pranswer](https://webrtc.github.io/samples/src/content/peerconnection/pr-answer/) * [约束和统计](https://webrtc.github.io/samples/src/content/peerconnection/constraints/) * [更多限制和统计](https://webrtc.github.io/samples/src/content/peerconnection/old-new-stats/) * [显示针对各种场景的createOffer输出](https://webrtc.github.io/samples/src/content/peerconnection/create-offer/) * [使用RTCDTMFSender](https://webrtc.github.io/samples/src/content/peerconnection/dtmf/) * [显示对等连接状态](https://webrtc.github.io/samples/src/content/peerconnection/states/) * [从STUN / TURN服务器收集ICE候选人](https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/) * [重新启动ICE](https://webrtc.github.io/samples/src/content/peerconnection/restart-ice/) * [Web音频输出作为对等连接的输入](https://webrtc.github.io/samples/src/content/peerconnection/webaudio-input/) * [对等连接作为Web音频的输入](https://webrtc.github.io/samples/src/content/peerconnection/webaudio-output/) * [使用可插入流进行端到端加密](https://webrtc.github.io/samples/src/content/peerconnection/endtoend-encryption) （实验性）* [使用可插入流的视频分析仪](https://webrtc.github.io/samples/src/content/peerconnection/video-analyzer) （实验性） ## [RTCDataChannel：](https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel) 通过对等连接发送任意数据 * [传送文字](https://webrtc.github.io/samples/src/content/datachannel/basic/) * [传送档案](https://webrtc.github.io/samples/src/content/datachannel/filetransfer/) * [传输资料](https://webrtc.github.io/samples/src/content/datachannel/datatransfer/) * [讯息传递](https://webrtc.github.io/samples/src/content/datachannel/messaging/)