19 changed files with 796 additions and 68 deletions
@ -1,41 +1,41 @@
|
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<template> |
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<div> |
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<video ref="localVideo" autoplay></video> |
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<video ref="remoteVideo" autoplay></video> |
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<button @click="startCall">Start Call</button> |
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<!-- 用于显示视频流的video元素 --> |
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<video ref="videoPlayer" id="videoPlayer" class="video-js vjs-default-skin" controls></video> |
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</div> |
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</template> |
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<script setup> |
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import { ref, onMounted } from 'vue'; |
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import WebRTCService from "@/api/mv/WebRTCService.js"; |
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<script> |
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// 导入 srs.js 库 |
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import 'https://cdn.jsdelivr.net/npm/srs.js@latest/dist/srs.min.js'; // 使用最新版本的 srs.js |
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const webrtcService = new WebRTCService(); |
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const localVideo = ref(null); |
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const remoteVideo = ref(null); |
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const localStream = ref(null); |
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const remoteStream = ref(null); |
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onMounted(async () => { |
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localStream.value = await webrtcService.init(); |
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localVideo.value.srcObject = localStream.value; |
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webrtcService.peer.ontrack = (event) => { |
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remoteStream.value = event.streams[0]; |
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remoteVideo.value.srcObject = remoteStream.value; |
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}; |
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export default { |
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mounted() { |
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// 初始化 WebRTC 播放器 |
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this.player = new SRS.Player({ |
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url: 'webrtc://47.121.143.167/live/livestream', // WebRTC流地址 |
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el: this.$refs.videoPlayer, // 视频播放元素 |
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width: '100%', // 宽度设置为 100% |
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height: 'auto', // 高度自适应 |
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}); |
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const startCall = () => { |
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webrtcService.createOffer(); |
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this.player.play().catch((err) => { |
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console.error('Failed to play WebRTC stream:', err); |
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}); |
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}, |
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beforeDestroy() { |
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// 销毁播放器 |
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if (this.player) { |
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this.player.dispose(); |
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} |
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} |
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}; |
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</script> |
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<style scoped> |
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video { |
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width: 45%; |
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/* 确保视频播放器自适应 */ |
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#videoPlayer { |
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width: 100%; |
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height: auto; |
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border: 1px solid #ccc; |
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margin: 10px; |
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} |
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</style> |
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@ -0,0 +1,718 @@
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//
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// Copyright (c) 2013-2021 Winlin
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//
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// SPDX-License-Identifier: MIT
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//
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'use strict'; |
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function SrsError(name, message) { |
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this.name = name; |
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this.message = message; |
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this.stack = (new Error()).stack; |
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} |
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SrsError.prototype = Object.create(Error.prototype); |
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SrsError.prototype.constructor = SrsError; |
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// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
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// Async-awat-prmise based SRS RTC Publisher.
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function SrsRtcPublisherAsync() { |
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var self = {}; |
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// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
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self.constraints = { |
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audio: true, |
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video: { |
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width: {ideal: 320, max: 576} |
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} |
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}; |
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// @see https://github.com/rtcdn/rtcdn-draft
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// @url The WebRTC url to play with, for example:
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// webrtc://r.ossrs.net/live/livestream
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// or specifies the API port:
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// webrtc://r.ossrs.net:11985/live/livestream
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// or autostart the publish:
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// webrtc://r.ossrs.net/live/livestream?autostart=true
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// or change the app from live to myapp:
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// webrtc://r.ossrs.net:11985/myapp/livestream
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// or change the stream from livestream to mystream:
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(eip) of answer:
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// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
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// or use plaintext, without SRTP:
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// webrtc://r.ossrs.net/live/livestream?encrypt=false
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// or any other information, will pass-by in the query:
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// webrtc://r.ossrs.net/live/livestream?vhost=xxx
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// webrtc://r.ossrs.net/live/livestream?token=xxx
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self.publish = async function (url) { |
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var conf = self.__internal.prepareUrl(url); |
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self.pc.addTransceiver("audio", {direction: "sendonly"}); |
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self.pc.addTransceiver("video", {direction: "sendonly"}); |
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//self.pc.addTransceiver("video", {direction: "sendonly"});
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//self.pc.addTransceiver("audio", {direction: "sendonly"});
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if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') { |
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throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`); |
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} |
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var stream = await navigator.mediaDevices.getUserMedia(self.constraints); |
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// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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stream.getTracks().forEach(function (track) { |
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self.pc.addTrack(track); |
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// Notify about local track when stream is ok.
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self.ontrack && self.ontrack({track: track}); |
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}); |
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var offer = await self.pc.createOffer(); |
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await self.pc.setLocalDescription(offer); |
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var session = await new Promise(function (resolve, reject) { |
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = { |
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api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl, |
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clientip: null, sdp: offer.sdp |
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}; |
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console.log("Generated offer: ", data); |
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const xhr = new XMLHttpRequest(); |
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xhr.onload = function() { |
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if (xhr.readyState !== xhr.DONE) return; |
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if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr); |
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const data = JSON.parse(xhr.responseText); |
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console.log("Got answer: ", data); |
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return data.code ? reject(xhr) : resolve(data); |
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} |
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xhr.open('POST', conf.apiUrl, true); |
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xhr.setRequestHeader('Content-type', 'application/json'); |
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xhr.send(JSON.stringify(data)); |
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}); |
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await self.pc.setRemoteDescription( |
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new RTCSessionDescription({type: 'answer', sdp: session.sdp}) |
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); |
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session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'; |
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return session; |
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}; |
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// Close the publisher.
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self.close = function () { |
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self.pc && self.pc.close(); |
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self.pc = null; |
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}; |
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// The callback when got local stream.
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// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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self.ontrack = function (event) { |
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// Add track to stream of SDK.
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self.stream.addTrack(event.track); |
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}; |
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// Internal APIs.
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self.__internal = { |
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defaultPath: '/rtc/v1/publish/', |
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prepareUrl: function (webrtcUrl) { |
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var urlObject = self.__internal.parse(webrtcUrl); |
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// If user specifies the schema, use it as API schema.
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var schema = urlObject.user_query.schema; |
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schema = schema ? schema + ':' : window.location.protocol; |
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var port = urlObject.port || 1985; |
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if (schema === 'https:') { |
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port = urlObject.port || 443; |
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} |
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// @see https://github.com/rtcdn/rtcdn-draft
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var api = urlObject.user_query.play || self.__internal.defaultPath; |
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if (api.lastIndexOf('/') !== api.length - 1) { |
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api += '/'; |
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} |
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var apiUrl = schema + '//' + urlObject.server + ':' + port + api; |
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for (var key in urlObject.user_query) { |
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if (key !== 'api' && key !== 'play') { |
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apiUrl += '&' + key + '=' + urlObject.user_query[key]; |
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} |
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} |
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// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
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apiUrl = apiUrl.replace(api + '&', api + '?'); |
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var streamUrl = urlObject.url; |
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return { |
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apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port, |
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tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7) |
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}; |
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}, |
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parse: function (url) { |
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// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
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var a = document.createElement("a"); |
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a.href = url.replace("rtmp://", "http://") |
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.replace("webrtc://", "http://") |
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.replace("rtc://", "http://"); |
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var vhost = a.hostname; |
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var app = a.pathname.substring(1, a.pathname.lastIndexOf("/")); |
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var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1); |
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// parse the vhost in the params of app, that srs supports.
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app = app.replace("...vhost...", "?vhost="); |
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if (app.indexOf("?") >= 0) { |
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var params = app.slice(app.indexOf("?")); |
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app = app.slice(0, app.indexOf("?")); |
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if (params.indexOf("vhost=") > 0) { |
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vhost = params.slice(params.indexOf("vhost=") + "vhost=".length); |
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if (vhost.indexOf("&") > 0) { |
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vhost = vhost.slice(0, vhost.indexOf("&")); |
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} |
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} |
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} |
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// when vhost equals to server, and server is ip,
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// the vhost is __defaultVhost__
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if (a.hostname === vhost) { |
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var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/; |
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if (re.test(a.hostname)) { |
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vhost = "__defaultVhost__"; |
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} |
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} |
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// parse the schema
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var schema = "rtmp"; |
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if (url.indexOf("://") > 0) { |
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schema = url.slice(0, url.indexOf("://")); |
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} |
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var port = a.port; |
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if (!port) { |
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// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
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if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) { |
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port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443; |
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} |
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// Guess by schema.
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if (schema === 'http') { |
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port = 80; |
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} else if (schema === 'https') { |
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port = 443; |
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} else if (schema === 'rtmp') { |
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port = 1935; |
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} |
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} |
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var ret = { |
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url: url, |
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schema: schema, |
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server: a.hostname, port: port, |
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vhost: vhost, app: app, stream: stream |
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}; |
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self.__internal.fill_query(a.search, ret); |
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// For webrtc API, we use 443 if page is https, or schema specified it.
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if (!ret.port) { |
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if (schema === 'webrtc' || schema === 'rtc') { |
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if (ret.user_query.schema === 'https') { |
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ret.port = 443; |
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} else if (window.location.href.indexOf('https://') === 0) { |
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ret.port = 443; |
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} else { |
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// For WebRTC, SRS use 1985 as default API port.
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ret.port = 1985; |
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} |
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} |
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} |
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return ret; |
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}, |
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fill_query: function (query_string, obj) { |
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// pure user query object.
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obj.user_query = {}; |
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if (query_string.length === 0) { |
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return; |
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} |
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// split again for angularjs.
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if (query_string.indexOf("?") >= 0) { |
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query_string = query_string.split("?")[1]; |
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} |
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var queries = query_string.split("&"); |
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for (var i = 0; i < queries.length; i++) { |
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var elem = queries[i]; |
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var query = elem.split("="); |
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obj[query[0]] = query[1]; |
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obj.user_query[query[0]] = query[1]; |
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} |
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// alias domain for vhost.
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if (obj.domain) { |
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obj.vhost = obj.domain; |
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} |
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} |
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}; |
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self.pc = new RTCPeerConnection(null); |
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// To keep api consistent between player and publisher.
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// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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// @see https://webrtc.org/getting-started/media-devices
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self.stream = new MediaStream(); |
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return self; |
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} |
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// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
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// Async-await-promise based SRS RTC Player.
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function SrsRtcPlayerAsync() { |
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var self = {}; |
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// @see https://github.com/rtcdn/rtcdn-draft
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// @url The WebRTC url to play with, for example:
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// webrtc://r.ossrs.net/live/livestream
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// or specifies the API port:
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// webrtc://r.ossrs.net:11985/live/livestream
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// webrtc://r.ossrs.net:80/live/livestream
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// or autostart the play:
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// webrtc://r.ossrs.net/live/livestream?autostart=true
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// or change the app from live to myapp:
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// webrtc://r.ossrs.net:11985/myapp/livestream
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// or change the stream from livestream to mystream:
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(eip) of answer:
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// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
|
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// or use plaintext, without SRTP:
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// webrtc://r.ossrs.net/live/livestream?encrypt=false
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// or any other information, will pass-by in the query:
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// webrtc://r.ossrs.net/live/livestream?vhost=xxx
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// webrtc://r.ossrs.net/live/livestream?token=xxx
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self.play = async function(url) { |
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var conf = self.__internal.prepareUrl(url); |
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self.pc.addTransceiver("audio", {direction: "recvonly"}); |
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self.pc.addTransceiver("video", {direction: "recvonly"}); |
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//self.pc.addTransceiver("video", {direction: "recvonly"});
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//self.pc.addTransceiver("audio", {direction: "recvonly"});
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var offer = await self.pc.createOffer(); |
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await self.pc.setLocalDescription(offer); |
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var session = await new Promise(function(resolve, reject) { |
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = { |
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api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl, |
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clientip: null, sdp: offer.sdp |
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}; |
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console.log("Generated offer: ", data); |
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const xhr = new XMLHttpRequest(); |
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xhr.onload = function() { |
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if (xhr.readyState !== xhr.DONE) return; |
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if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr); |
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const data = JSON.parse(xhr.responseText); |
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console.log("Got answer: ", data); |
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return data.code ? reject(xhr) : resolve(data); |
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} |
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xhr.open('POST', conf.apiUrl, true); |
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xhr.setRequestHeader('Content-type', 'application/json'); |
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xhr.send(JSON.stringify(data)); |
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}); |
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await self.pc.setRemoteDescription( |
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new RTCSessionDescription({type: 'answer', sdp: session.sdp}) |
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); |
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session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'; |
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return session; |
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}; |
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// Close the player.
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self.close = function() { |
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self.pc && self.pc.close(); |
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self.pc = null; |
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}; |
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// The callback when got remote track.
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// Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
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self.ontrack = function (event) { |
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// https://webrtc.org/getting-started/remote-streams
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self.stream.addTrack(event.track); |
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}; |
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// Internal APIs.
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self.__internal = { |
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defaultPath: '/rtc/v1/play/', |
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prepareUrl: function (webrtcUrl) { |
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var urlObject = self.__internal.parse(webrtcUrl); |
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// If user specifies the schema, use it as API schema.
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var schema = urlObject.user_query.schema; |
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schema = schema ? schema + ':' : window.location.protocol; |
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var port = urlObject.port || 1985; |
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if (schema === 'https:') { |
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port = urlObject.port || 443; |
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} |
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|
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// @see https://github.com/rtcdn/rtcdn-draft
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var api = urlObject.user_query.play || self.__internal.defaultPath; |
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if (api.lastIndexOf('/') !== api.length - 1) { |
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api += '/'; |
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} |
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var apiUrl = schema + '//' + urlObject.server + ':' + port + api; |
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for (var key in urlObject.user_query) { |
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if (key !== 'api' && key !== 'play') { |
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apiUrl += '&' + key + '=' + urlObject.user_query[key]; |
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} |
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} |
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// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
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apiUrl = apiUrl.replace(api + '&', api + '?'); |
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|
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var streamUrl = urlObject.url; |
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return { |
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apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port, |
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tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7) |
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}; |
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}, |
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parse: function (url) { |
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// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
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var a = document.createElement("a"); |
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a.href = url.replace("rtmp://", "http://") |
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.replace("webrtc://", "http://") |
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.replace("rtc://", "http://"); |
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|
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var vhost = a.hostname; |
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var app = a.pathname.substring(1, a.pathname.lastIndexOf("/")); |
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var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1); |
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|
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// parse the vhost in the params of app, that srs supports.
|
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app = app.replace("...vhost...", "?vhost="); |
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if (app.indexOf("?") >= 0) { |
||||
var params = app.slice(app.indexOf("?")); |
||||
app = app.slice(0, app.indexOf("?")); |
||||
|
||||
if (params.indexOf("vhost=") > 0) { |
||||
vhost = params.slice(params.indexOf("vhost=") + "vhost=".length); |
||||
if (vhost.indexOf("&") > 0) { |
||||
vhost = vhost.slice(0, vhost.indexOf("&")); |
||||
} |
||||
} |
||||
} |
||||
|
||||
// when vhost equals to server, and server is ip,
|
||||
// the vhost is __defaultVhost__
|
||||
if (a.hostname === vhost) { |
||||
var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/; |
||||
if (re.test(a.hostname)) { |
||||
vhost = "__defaultVhost__"; |
||||
} |
||||
} |
||||
|
||||
// parse the schema
|
||||
var schema = "rtmp"; |
||||
if (url.indexOf("://") > 0) { |
||||
schema = url.slice(0, url.indexOf("://")); |
||||
} |
||||
|
||||
var port = a.port; |
||||
if (!port) { |
||||
// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
|
||||
if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) { |
||||
port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443; |
||||
} |
||||
|
||||
// Guess by schema.
|
||||
if (schema === 'http') { |
||||
port = 80; |
||||
} else if (schema === 'https') { |
||||
port = 443; |
||||
} else if (schema === 'rtmp') { |
||||
port = 1935; |
||||
} |
||||
} |
||||
|
||||
var ret = { |
||||
url: url, |
||||
schema: schema, |
||||
server: a.hostname, port: port, |
||||
vhost: vhost, app: app, stream: stream |
||||
}; |
||||
self.__internal.fill_query(a.search, ret); |
||||
|
||||
// For webrtc API, we use 443 if page is https, or schema specified it.
|
||||
if (!ret.port) { |
||||
if (schema === 'webrtc' || schema === 'rtc') { |
||||
if (ret.user_query.schema === 'https') { |
||||
ret.port = 443; |
||||
} else if (window.location.href.indexOf('https://') === 0) { |
||||
ret.port = 443; |
||||
} else { |
||||
// For WebRTC, SRS use 1985 as default API port.
|
||||
ret.port = 1985; |
||||
} |
||||
} |
||||
} |
||||
|
||||
return ret; |
||||
}, |
||||
fill_query: function (query_string, obj) { |
||||
// pure user query object.
|
||||
obj.user_query = {}; |
||||
|
||||
if (query_string.length === 0) { |
||||
return; |
||||
} |
||||
|
||||
// split again for angularjs.
|
||||
if (query_string.indexOf("?") >= 0) { |
||||
query_string = query_string.split("?")[1]; |
||||
} |
||||
|
||||
var queries = query_string.split("&"); |
||||
for (var i = 0; i < queries.length; i++) { |
||||
var elem = queries[i]; |
||||
|
||||
var query = elem.split("="); |
||||
obj[query[0]] = query[1]; |
||||
obj.user_query[query[0]] = query[1]; |
||||
} |
||||
|
||||
// alias domain for vhost.
|
||||
if (obj.domain) { |
||||
obj.vhost = obj.domain; |
||||
} |
||||
} |
||||
}; |
||||
|
||||
self.pc = new RTCPeerConnection(null); |
||||
|
||||
// Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
|
||||
self.stream = new MediaStream(); |
||||
|
||||
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
|
||||
self.pc.ontrack = function(event) { |
||||
if (self.ontrack) { |
||||
self.ontrack(event); |
||||
} |
||||
}; |
||||
|
||||
return self; |
||||
} |
||||
|
||||
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
|
||||
// Async-awat-prmise based SRS RTC Publisher by WHIP.
|
||||
function SrsRtcWhipWhepAsync() { |
||||
var self = {}; |
||||
|
||||
// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
|
||||
self.constraints = { |
||||
audio: true, |
||||
video: { |
||||
width: {ideal: 320, max: 576} |
||||
} |
||||
}; |
||||
|
||||
// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
|
||||
// @url The WebRTC url to publish with, for example:
|
||||
// http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
|
||||
// @options The options to control playing, supports:
|
||||
// camera: boolean, whether capture video from camera, default to true.
|
||||
// screen: boolean, whether capture video from screen, default to false.
|
||||
// audio: boolean, whether play audio, default to true.
|
||||
self.publish = async function (url, options) { |
||||
if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`); |
||||
const hasAudio = options?.audio ?? true; |
||||
const useCamera = options?.camera ?? true; |
||||
const useScreen = options?.screen ?? false; |
||||
|
||||
if (!hasAudio && !useCamera && !useScreen) throw new Error(`The camera, screen and audio can't be false at the same time`); |
||||
|
||||
if (hasAudio) { |
||||
self.pc.addTransceiver("audio", {direction: "sendonly"}); |
||||
} else { |
||||
self.constraints.audio = false; |
||||
} |
||||
|
||||
if (useCamera || useScreen) { |
||||
self.pc.addTransceiver("video", {direction: "sendonly"}); |
||||
} |
||||
|
||||
if (!useCamera) { |
||||
self.constraints.video = false; |
||||
} |
||||
|
||||
if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') { |
||||
throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`); |
||||
} |
||||
|
||||
if (useScreen) { |
||||
const displayStream = await navigator.mediaDevices.getDisplayMedia({ |
||||
video: true |
||||
}); |
||||
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
|
||||
displayStream.getTracks().forEach(function (track) { |
||||
self.pc.addTrack(track); |
||||
// Notify about local track when stream is ok.
|
||||
self.ontrack && self.ontrack({track: track}); |
||||
}); |
||||
} |
||||
|
||||
if (useCamera || hasAudio) { |
||||
const userStream = await navigator.mediaDevices.getUserMedia(self.constraints); |
||||
|
||||
userStream.getTracks().forEach(function (track) { |
||||
self.pc.addTrack(track); |
||||
// Notify about local track when stream is ok.
|
||||
self.ontrack && self.ontrack({track: track}); |
||||
}); |
||||
} |
||||
|
||||
var offer = await self.pc.createOffer(); |
||||
await self.pc.setLocalDescription(offer); |
||||
const answer = await new Promise(function (resolve, reject) { |
||||
console.log(`Generated offer: ${offer.sdp}`); |
||||
|
||||
const xhr = new XMLHttpRequest(); |
||||
xhr.onload = function() { |
||||
if (xhr.readyState !== xhr.DONE) return; |
||||
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr); |
||||
const data = xhr.responseText; |
||||
console.log("Got answer: ", data); |
||||
return data.code ? reject(xhr) : resolve(data); |
||||
} |
||||
xhr.open('POST', url, true); |
||||
xhr.setRequestHeader('Content-type', 'application/sdp'); |
||||
xhr.send(offer.sdp); |
||||
}); |
||||
await self.pc.setRemoteDescription( |
||||
new RTCSessionDescription({type: 'answer', sdp: answer}) |
||||
); |
||||
|
||||
return self.__internal.parseId(url, offer.sdp, answer); |
||||
}; |
||||
|
||||
// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
|
||||
// @url The WebRTC url to play with, for example:
|
||||
// http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
|
||||
// @options The options to control playing, supports:
|
||||
// videoOnly: boolean, whether only play video, default to false.
|
||||
// audioOnly: boolean, whether only play audio, default to false.
|
||||
self.play = async function(url, options) { |
||||
if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`); |
||||
if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`); |
||||
|
||||
if (!options?.videoOnly) self.pc.addTransceiver("audio", {direction: "recvonly"}); |
||||
if (!options?.audioOnly) self.pc.addTransceiver("video", {direction: "recvonly"}); |
||||
|
||||
var offer = await self.pc.createOffer(); |
||||
await self.pc.setLocalDescription(offer); |
||||
const answer = await new Promise(function(resolve, reject) { |
||||
console.log(`Generated offer: ${offer.sdp}`); |
||||
|
||||
const xhr = new XMLHttpRequest(); |
||||
xhr.onload = function() { |
||||
if (xhr.readyState !== xhr.DONE) return; |
||||
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr); |
||||
const data = xhr.responseText; |
||||
console.log("Got answer: ", data); |
||||
return data.code ? reject(xhr) : resolve(data); |
||||
} |
||||
xhr.open('POST', url, true); |
||||
xhr.setRequestHeader('Content-type', 'application/sdp'); |
||||
xhr.send(offer.sdp); |
||||
}); |
||||
await self.pc.setRemoteDescription( |
||||
new RTCSessionDescription({type: 'answer', sdp: answer}) |
||||
); |
||||
|
||||
return self.__internal.parseId(url, offer.sdp, answer); |
||||
}; |
||||
|
||||
// Close the publisher.
|
||||
self.close = function () { |
||||
self.pc && self.pc.close(); |
||||
self.pc = null; |
||||
}; |
||||
|
||||
// The callback when got local stream.
|
||||
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
|
||||
self.ontrack = function (event) { |
||||
// Add track to stream of SDK.
|
||||
self.stream.addTrack(event.track); |
||||
}; |
||||
|
||||
self.pc = new RTCPeerConnection(null); |
||||
|
||||
// To keep api consistent between player and publisher.
|
||||
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
|
||||
// @see https://webrtc.org/getting-started/media-devices
|
||||
self.stream = new MediaStream(); |
||||
|
||||
// Internal APIs.
|
||||
self.__internal = { |
||||
parseId: (url, offer, answer) => { |
||||
let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length); |
||||
sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':'; |
||||
sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length); |
||||
sessionid = sessionid.substr(0, sessionid.indexOf('\n')); |
||||
|
||||
const a = document.createElement("a"); |
||||
a.href = url; |
||||
return { |
||||
sessionid: sessionid, // Should be ice-ufrag of answer:offer.
|
||||
simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/', |
||||
}; |
||||
}, |
||||
}; |
||||
|
||||
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
|
||||
self.pc.ontrack = function(event) { |
||||
if (self.ontrack) { |
||||
self.ontrack(event); |
||||
} |
||||
}; |
||||
|
||||
return self; |
||||
} |
||||
|
||||
// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
|
||||
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
|
||||
function SrsRtcFormatSenders(senders, kind) { |
||||
var codecs = []; |
||||
senders.forEach(function (sender) { |
||||
var params = sender.getParameters(); |
||||
params && params.codecs && params.codecs.forEach(function(c) { |
||||
if (kind && sender.track.kind !== kind) { |
||||
return; |
||||
} |
||||
|
||||
if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) { |
||||
return; |
||||
} |
||||
|
||||
var s = ''; |
||||
|
||||
s += c.mimeType.replace('audio/', '').replace('video/', ''); |
||||
s += ', ' + c.clockRate + 'HZ'; |
||||
if (sender.track.kind === "audio") { |
||||
s += ', channels: ' + c.channels; |
||||
} |
||||
s += ', pt: ' + c.payloadType; |
||||
|
||||
codecs.push(s); |
||||
}); |
||||
}); |
||||
return codecs.join(", "); |
||||
} |
||||
export default SrsRtcPlayerAsync |
||||
Loading…
Reference in new issue