diff --git a/src/assets/js/srs.sdk.js b/src/assets/js/srs.sdk.js new file mode 100644 index 00000000..87e20194 --- /dev/null +++ b/src/assets/js/srs.sdk.js @@ -0,0 +1,707 @@ +// +// Copyright (c) 2013-2021 Winlin +// +// SPDX-License-Identifier: MIT +// + +"use strict"; + +function SrsError(name, message) { + this.name = name; + this.message = message; + this.stack = new Error().stack; +} +SrsError.prototype = Object.create(Error.prototype); +SrsError.prototype.constructor = SrsError; + +// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter +// Async-awat-prmise based SRS RTC Publisher. +function SrsRtcPublisherAsync() { + 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://github.com/rtcdn/rtcdn-draft + // @url The WebRTC url to play with, for example: + // webrtc://r.ossrs.net/live/livestream + // or specifies the API port: + // webrtc://r.ossrs.net:11985/live/livestream + // or autostart the publish: + // webrtc://r.ossrs.net/live/livestream?autostart=true + // or change the app from live to myapp: + // webrtc://r.ossrs.net:11985/myapp/livestream + // or change the stream from livestream to mystream: + // webrtc://r.ossrs.net:11985/live/mystream + // or set the api server to myapi.domain.com: + // webrtc://myapi.domain.com/live/livestream + // or set the candidate(eip) of answer: + // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185 + // or force to access https API: + // webrtc://r.ossrs.net/live/livestream?schema=https + // or use plaintext, without SRTP: + // webrtc://r.ossrs.net/live/livestream?encrypt=false + // or any other information, will pass-by in the query: + // webrtc://r.ossrs.net/live/livestream?vhost=xxx + // webrtc://r.ossrs.net/live/livestream?token=xxx + self.publish = async function (url) { + var conf = self.__internal.prepareUrl(url); + self.pc.addTransceiver("audio", { direction: "sendonly" }); + self.pc.addTransceiver("video", { direction: "sendonly" }); + //self.pc.addTransceiver("video", {direction: "sendonly"}); + //self.pc.addTransceiver("audio", {direction: "sendonly"}); + + 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` + ); + } + var stream = await navigator.mediaDevices.getUserMedia(self.constraints); + + // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack + stream.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); + var session = await new Promise(function (resolve, reject) { + // @see https://github.com/rtcdn/rtcdn-draft + var data = { + api: conf.apiUrl, + tid: conf.tid, + streamurl: conf.streamUrl, + clientip: null, + sdp: offer.sdp + }; + console.log("Generated offer: ", data); + + 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 = JSON.parse(xhr.responseText); + console.log("Got answer: ", data); + return data.code ? reject(xhr) : resolve(data); + }; + xhr.open("POST", conf.apiUrl, true); + xhr.setRequestHeader("Content-type", "application/json"); + xhr.send(JSON.stringify(data)); + }); + await self.pc.setRemoteDescription(new RTCSessionDescription({ type: "answer", sdp: session.sdp })); + session.simulator = conf.schema + "//" + conf.urlObject.server + ":" + conf.port + "/rtc/v1/nack/"; + + return session; + }; + + // 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); + }; + + // Internal APIs. + self.__internal = { + defaultPath: "/rtc/v1/publish/", + prepareUrl: function (webrtcUrl) { + var urlObject = self.__internal.parse(webrtcUrl); + + // If user specifies the schema, use it as API schema. + var schema = urlObject.user_query.schema; + schema = schema ? schema + ":" : window.location.protocol; + + var port = urlObject.port || 1985; + if (schema === "https:") { + port = urlObject.port || 443; + } + + // @see https://github.com/rtcdn/rtcdn-draft + var api = urlObject.user_query.play || self.__internal.defaultPath; + if (api.lastIndexOf("/") !== api.length - 1) { + api += "/"; + } + + var apiUrl = schema + "//" + urlObject.server + ":" + port + api; + for (var key in urlObject.user_query) { + if (key !== "api" && key !== "play") { + apiUrl += "&" + key + "=" + urlObject.user_query[key]; + } + } + // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v + apiUrl = apiUrl.replace(api + "&", api + "?"); + + var streamUrl = urlObject.url; + + return { + apiUrl: apiUrl, + streamUrl: streamUrl, + schema: schema, + urlObject: urlObject, + port: port, + tid: Number(parseInt(new Date().getTime() * Math.random() * 100)) + .toString(16) + .slice(0, 7) + }; + }, + parse: function (url) { + // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri + var a = document.createElement("a"); + a.href = url.replace("rtmp://", "http://").replace("webrtc://", "http://").replace("rtc://", "http://"); + + var vhost = a.hostname; + var app = a.pathname.substring(1, a.pathname.lastIndexOf("/")); + var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1); + + // parse the vhost in the params of app, that srs supports. + app = app.replace("...vhost...", "?vhost="); + 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); + + // 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(); + + return self; +} + +// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter +// Async-await-promise based SRS RTC Player. +function SrsRtcPlayerAsync() { + var self = {}; + + // @see https://github.com/rtcdn/rtcdn-draft + // @url The WebRTC url to play with, for example: + // webrtc://r.ossrs.net/live/livestream + // or specifies the API port: + // webrtc://r.ossrs.net:11985/live/livestream + // webrtc://r.ossrs.net:80/live/livestream + // or autostart the play: + // webrtc://r.ossrs.net/live/livestream?autostart=true + // or change the app from live to myapp: + // webrtc://r.ossrs.net:11985/myapp/livestream + // or change the stream from livestream to mystream: + // webrtc://r.ossrs.net:11985/live/mystream + // or set the api server to myapi.domain.com: + // webrtc://myapi.domain.com/live/livestream + // or set the candidate(eip) of answer: + // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185 + // or force to access https API: + // webrtc://r.ossrs.net/live/livestream?schema=https + // or use plaintext, without SRTP: + // webrtc://r.ossrs.net/live/livestream?encrypt=false + // or any other information, will pass-by in the query: + // webrtc://r.ossrs.net/live/livestream?vhost=xxx + // webrtc://r.ossrs.net/live/livestream?token=xxx + self.play = async function (url) { + var conf = self.__internal.prepareUrl(url); + self.pc.addTransceiver("audio", { direction: "recvonly" }); + self.pc.addTransceiver("video", { direction: "recvonly" }); + //self.pc.addTransceiver("video", {direction: "recvonly"}); + //self.pc.addTransceiver("audio", {direction: "recvonly"}); + + var offer = await self.pc.createOffer(); + await self.pc.setLocalDescription(offer); + var session = await new Promise(function (resolve, reject) { + // @see https://github.com/rtcdn/rtcdn-draft + var data = { + api: conf.apiUrl, + tid: conf.tid, + streamurl: conf.streamUrl, + clientip: null, + sdp: offer.sdp + }; + console.log("Generated offer: ", data); + + 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 = JSON.parse(xhr.responseText); + console.log("Got answer: ", data); + return data.code ? reject(xhr) : resolve(data); + }; + xhr.open("POST", conf.apiUrl, true); + xhr.setRequestHeader("Content-type", "application/json"); + xhr.send(JSON.stringify(data)); + }); + await self.pc.setRemoteDescription(new RTCSessionDescription({ type: "answer", sdp: session.sdp })); + session.simulator = conf.schema + "//" + conf.urlObject.server + ":" + conf.port + "/rtc/v1/nack/"; + + return session; + }; + + // Close the player. + self.close = function () { + self.pc && self.pc.close(); + self.pc = null; + }; + + // The callback when got remote track. + // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream + self.ontrack = function (event) { + // https://webrtc.org/getting-started/remote-streams + self.stream.addTrack(event.track); + }; + + // Internal APIs. + self.__internal = { + defaultPath: "/rtc/v1/play/", + prepareUrl: function (webrtcUrl) { + var urlObject = self.__internal.parse(webrtcUrl); + + // If user specifies the schema, use it as API schema. + var schema = urlObject.user_query.schema; + schema = schema ? schema + ":" : window.location.protocol; + + var port = urlObject.port || 1985; + if (schema === "https:") { + port = urlObject.port || 443; + } + + // @see https://github.com/rtcdn/rtcdn-draft + var api = urlObject.user_query.play || self.__internal.defaultPath; + if (api.lastIndexOf("/") !== api.length - 1) { + api += "/"; + } + + var apiUrl = schema + "//" + urlObject.server + ":" + port + api; + for (var key in urlObject.user_query) { + if (key !== "api" && key !== "play") { + apiUrl += "&" + key + "=" + urlObject.user_query[key]; + } + } + // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v + apiUrl = apiUrl.replace(api + "&", api + "?"); + + var streamUrl = urlObject.url; + + return { + apiUrl: apiUrl, + streamUrl: streamUrl, + schema: schema, + urlObject: urlObject, + port: port, + tid: Number(parseInt(new Date().getTime() * Math.random() * 100)) + .toString(16) + .slice(0, 7) + }; + }, + parse: function (url) { + // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri + var a = document.createElement("a"); + a.href = url.replace("rtmp://", "http://").replace("webrtc://", "http://").replace("rtc://", "http://"); + + var vhost = a.hostname; + var app = a.pathname.substring(1, a.pathname.lastIndexOf("/")); + var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1); + + // parse the vhost in the params of app, that srs supports. + app = app.replace("...vhost...", "?vhost="); + 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 + self.publish = async function (url) { + if (url.indexOf("/whip/") === -1) throw new Error(`invalid WHIP url ${url}`); + + self.pc.addTransceiver("audio", { direction: "sendonly" }); + self.pc.addTransceiver("video", { direction: "sendonly" }); + + 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` + ); + } + var stream = await navigator.mediaDevices.getUserMedia(self.constraints); + + // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack + stream.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); + + 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 + self.play = async function (url) { + if (url.indexOf("/whip-play/") === -1 && url.indexOf("/whep/") === -1) throw new Error(`invalid WHEP url ${url}`); + + self.pc.addTransceiver("audio", { direction: "recvonly" }); + 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); + + 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 { + SrsRtcPublisherAsync, + SrsRtcPlayerAsync, + SrsRtcWhipWhepAsync, + SrsRtcFormatSenders, + SrsError +}; diff --git a/src/router/index.js b/src/router/index.js index 8a5f29ca..cbc7b1d6 100644 --- a/src/router/index.js +++ b/src/router/index.js @@ -3034,6 +3034,13 @@ const routes2 = [ component: (resolve) => require(["@/views/projectFront/insideVideo/overview.vue"], resolve), }, + // 内网视频(srs) + { + path: "/project/insideSrsVideo", + name: "insideSrsVideo", + component: (resolve) => + require(["@/views/projectFront/insideSrsVideo/overview.vue"], resolve), + }, { path: "/project/imgRanging", name: "imgRanging", diff --git a/src/views/projectFront/insideSrsVideo/ckplayerComp.vue b/src/views/projectFront/insideSrsVideo/ckplayerComp.vue new file mode 100644 index 00000000..d2268008 --- /dev/null +++ b/src/views/projectFront/insideSrsVideo/ckplayerComp.vue @@ -0,0 +1,34 @@ + + + + diff --git a/src/views/projectFront/insideSrsVideo/overview.vue b/src/views/projectFront/insideSrsVideo/overview.vue new file mode 100644 index 00000000..0a13c73d --- /dev/null +++ b/src/views/projectFront/insideSrsVideo/overview.vue @@ -0,0 +1,494 @@ + + + diff --git a/src/views/projectFront/insideSrsVideo/style.less b/src/views/projectFront/insideSrsVideo/style.less new file mode 100644 index 00000000..35cf0850 --- /dev/null +++ b/src/views/projectFront/insideSrsVideo/style.less @@ -0,0 +1,14 @@ +.videoOverview{ + background-color: #0D1534; +} +.header{ + background: url('../../../assets/images/overview3/headerBG.png') center; + height: 70px; + width: 100%; +} +.videoContent{ + display: flex; + height: calc(100% - 70px); + // background-color: #2FBBEC; + overflow: hidden; +}