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fix: BUG修改

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This commit is contained in:
kun 2024-04-17 21:57:06 +08:00
parent 760e30013c
commit 881baac20e
6 changed files with 825 additions and 14 deletions
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3
.env.development
View File

@ -15,12 +15,13 @@ NODE_ENV = 'development'
# 演示平台
# VITE_API_URL = 'http://jxj.zhgdyun.com:9809'
# 百色七参数线上地址
VITE_API_URL = 'http://101.43.164.214:11111'
# VITE_API_URL = 'http://101.43.164.214:11111'
# 七参数标准版(测试平台)
# VITE_API_URL = 'http://jxj.zhgdyun.com:15551'
# 七参数标准版(测试平台)
# VITE_API_URL = 'http://182.90.224.237:15551'
# agjt
VITE_API_URL = 'http://jxj.zhgdyun.com:19814'
# VITE_API_URL = 'http://47.93.215.234:9809'
# 上传
VITE_ULD_API_URL = 'http://192.168.34.155:8012/onlinePreview?url='

4
.env.production
View File

@ -12,7 +12,7 @@ NODE_ENV = "production"
# VITE_API_URL = "http://183.249.224.118:9003"
# 百色 新项目通用地址
VITE_API_URL = 'http://101.43.164.214:11111'
# VITE_API_URL = 'http://101.43.164.214:11111'
# 七参数标准版(演示平台)
# VITE_API_URL = 'http://jxj.zhgdyun.com:9809'
@ -21,7 +21,7 @@ VITE_API_URL = 'http://101.43.164.214:11111'
# 七参数标准版(测试平台)
# VITE_API_URL = 'http://182.90.224.237:15551'
# agjt
# VITE_API_URL = 'http://47.93.215.234:9809'
VITE_API_URL = 'http://47.93.215.234:9809'
# 打包
VITE_ULD_API_URL = 'http://jxj.zhgdyun.com:8012/onlinePreview?url='

5
src/api/modules/video.ts
View File

@ -14,3 +14,8 @@ export const selectUserVideoListApi = (params: {}) => {
export const getSafeHatSessionApi = (params: {}) => {
return http.post(BASEURL + `/xmgl/projectApi/getSafeyHatSession`, params);
};
// 现场视频
export const selectLiveVideoListApi = (params: {}) => {
return http.get(BASEURL + `/xmgl/project/getProjectExtendInfo`, params);
};

707
src/assets/js/srs.sdk.js Normal file
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@ -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
};

77
src/views/sevenLargeScreen/comprehensiveManage/projectOverview/centerTop.vue
View File

@ -12,7 +12,21 @@
{{ item.title }}
</div>
</div>
<div class="videoBox" v-if="showVideo" @mouseenter="showChangeVideo = true" @mouseleave="showChangeVideo = false">
<div class="href-content" v-if="showVideo == 1">
<el-carousel indicator-position="none" height="450px" style="width: 76%;">
<el-carousel-item v-for="(item, index) in videoList" :key="item.id">
<ckplayerComp
:name="index"
:poster="''"
:deviceIp="`http://${item.account}:${item.password}`"
:videoUrls="item.serialNumber"
:autoPlay="true"
></ckplayerComp>
</el-carousel-item>
</el-carousel>
</div>
<div class="videoBox" v-if="showVideo == 2" @mouseenter="showChangeVideo = true" @mouseleave="showChangeVideo = false">
<el-upload
:action="BASEURL + '/upload/image'"
:on-success="file => handleSuccessTwo(file, 1)"
@ -26,7 +40,7 @@
</el-upload>
<video :src="BASEURL + '/image/' + projectData.videoUrl" class="videos" autoplay controls loop></video>
</div>
<div class="imgBox" v-else>
<div class="imgBox" v-if="showVideo == 3">
<div class="imgs" @mouseenter="showChangeImg = true" @mouseleave="showChangeImg = false">
<el-upload
:action="BASEURL + '/upload/image'"
@ -53,8 +67,11 @@ import { GlobalStore } from "@/stores";
import { ElMessage } from "element-plus";
import { editProjectInfo, eidtProjectShowConfig, queryBySnData } from "@/api/modules/projectOverview";
import { selectLiveVideoListApi } from "@/api/modules/video";
import ckplayerComp from "./ckplayerComp.vue";
const store = GlobalStore();
const videoList = ref([] as any);
// ts
type Props = {
projectData?: any; // 传入项目信息
@ -81,7 +98,7 @@ const picUrl = ref("" as any);
const BASEURL = import.meta.env.VITE_API_URL;
// 显示视频
const showVideo = ref(true as any);
const showVideo = ref(1 as any);
// 显示更换图片
const showChangeImg = ref(false as any);
@ -90,9 +107,32 @@ const showChangeImg = ref(false as any);
const showChangeVideo = ref(false as any);
let topText = ref([
{ id: 1, title: "宣传视频", isActive: true },
{ id: 2, title: "效果图", isActive: false }
{ id: 1, title: "现场视频", isActive: true },
{ id: 2, title: "宣传视频", isActive: false },
{ id: 3, title: "效果图", isActive: false }
]);
const getVideoList = async () => {
let res: any = await selectLiveVideoListApi({
projectSn: store.sn
});
console.log(res,'445566')
if(res.result && res.result.extend1){
videoList.value = JSON.parse(res.result.extend1).result.videoList;
console.log(videoList.value,'112233')
// 为了解决视频播放器渲染，第二个总是会默认显示一半，我动态设置样式让视图刷新，只要设置百分百就有问题，所以只能使用此方法
setTimeout(() => {
// 获取所有的 video 元素
var videos = document.querySelectorAll(".href-content video")
// var videos = document.getElementsByTagName("video");
// 遍历所有的 video 元素
for (var i = 0; i < videos.length; i++) {
// 修改视频元素的样式
videos[i].style.width = "99.9%";
videos[i].style.height = "99.9%";
}
}, 2000);
}
};
function boxStyle(item: any) {
if (item.isActive) {
@ -113,11 +153,8 @@ function activeBtn(item: any) {
item.isActive = !currentState;
tabIndex.value = item.id;
}
if (item.id === 1) {
showVideo.value = true;
} else {
showVideo.value = false;
}
showVideo.value = item.id;
console.log(showVideo.value)
}
const uploadFail = () => {
@ -235,7 +272,8 @@ function saveOrDeleteVideo(url) {
defineExpose({
getQueryBySnData
})
onMounted(() => {
onMounted(async () => {
await getVideoList();
getQueryBySnData();
});
</script>
@ -275,8 +313,25 @@ onMounted(() => {
}
}
}
.href-content {
width: 100%;
height: 92%;
margin-top: 8%;
background: url("@/assets/images/comprehensiveManage/project10.png") no-repeat;
background-size: 100% 100%;
display: flex;
align-items: center;
justify-content: center;
}
}
// .href-content {
// width: 95%;
// height: 92%;
// margin: 0 auto;
// margin-top: 8%;
// }
.change-video {
position: absolute;
background: url("@/assets/images/cardImg.png") no-repeat;

43
src/views/sevenLargeScreen/comprehensiveManage/projectOverview/ckplayerComp.vue Normal file
View File

@ -0,0 +1,43 @@
<!--ckplayer 视频播放-->
<template>
<div style="width: 100%;height: 100%">
<video :id="'videoItem' + name" muted controls autoplay style="width: 100%;height: 100%;object-fit: fill;" />
</div>
</template>
<script>
import Srs from "@/assets/js/srs.sdk";
export default {
props: ["videoUrls", "autoPlay", "poster", "deviceIp", "name"],
data() {
return {
player: "",
rtcPlayer: null
};
},
mounted: function () {
console.log(this.videoUrls, this.deviceIp, this.autoPlay, this.name, 666);
this.player = document.getElementById("videoItem" + this.name);
//方法一：使用srs.sdk.js
this.rtcPlayer = new Srs.SrsRtcWhipWhepAsync();
this.rtcPlayer.play(this.videoUrls);
// video标签
this.player.srcObject = this.rtcPlayer.stream;
// 方法二：使用jswebrtc.min.js
// new JSWebrtc.Player(this.url, { video: player, autoplay: true, });
},
beforeDestroy: function () {
this.rtcPlayer.close();
},
watch: {
// watch第一次绑定值的时候不会执行监听，修改数据才会触发函数
videoUrls(newVal,oldVal) {
console.log(newVal)
this.rtcPlayer = new Srs.SrsRtcWhipWhepAsync();
this.rtcPlayer.play(newVal);
this.player.srcObject = this.rtcPlayer.stream;
}
},
methods: {}
};
</script>