一、效果图
二、主要功能
1. 创建自己的语音直播间
2. 查询所有直播间列表
3.加入房间
4.申请上位
5.麦克风控制
6.声音控制
7.赠送礼物（特效 + 批量移动动画）
8.退出房间
三、原理
1.uniapp 实现客户端H5、安卓、苹果
2.webRTC实现语音直播间（具体原理网上有很多文章我就不讲了，贴个图）
3.使用node.js搭建信令服务器（我用的是socket）
4.礼物及特效使用svga
四、踩坑及解决方案
1. 客户端（这里重点在于app端）一定要在视图层创建webRTC！！！不要在逻辑层创建！！！因为会要求使用安全连接，也就是说要用到SSL证书，这个很多人都没有，有的话当我没说。如何在视图层创建RTC呢？在uniapp中使用renderjs！
<script module="webRTC" lang="renderjs">
new RTCPeerConnection(iceServers)
</script>
2. （这里重点也在于app）客户端创建和信令服务器进行通信的socket时app端在页面跳转后socket状态消失无法响应信令服务器消息。解决方案是：一定不要在客户端视图层创建socket！！！也就是说socket不要创建在renderjs里，要在逻辑层用uniapp提供的api进行创建，然后使用uniapp文档中说明的逻辑层和视图层的通信方式进行通信，这样虽然在开发中有些繁琐，但是能解决问题。
onShow(){
// socketTask是使用uniapp提供的uni.connectSocket创建出来的socket实例
// watchSocketMessage代理了socket实例的onMessage方法
socketTask.watchSocketMessage = (data) => {
this.watchSocketMessage(data)
}
}
methed:{
watchSocketMessage(){
// 这里是收到信令服务器socket后的逻辑
}
}// 这里是逻辑层和renderjs通信的方式，通过监听状态的改变从而触发renderjs的对应的方法
// 注意在页面刚加载完成后这些方法会被默认触发一边，所以要在这些放方法做好判断return出去
<view :rid="rid" :change:rid="webRTC.initRid" :userId="userId" :change:userId="webRTC.initUserId"
:giftnum="giftnum" :change:giftnum="webRTC.initgiftnum" :micPosition="micPosition"
:change:micPosition="webRTC.initMicPositions" :giftPosition="giftPosition"
:change:giftPosition="webRTC.initGiftPosition" :RTCJoin="RTCJoin" :change:RTCJoin="webRTC.changeRTCjoin"
:RTCOffier="RTCOffier" :change:RTCOffier="webRTC.changeRTCoffier" :RTCAnswer="RTCAnswer" :isAudio="isAudio"
:change:isAudio="webRTC.changeIsAudio" :change:RTCAnswer="webRTC.changeRTCAnswer"
:RT***andidate="RT***andidate" :change:RT***andidate="webRTC.changeRT***andidate" :isTrue="isTrue"
:change:isTrue="webRTC.changeIsTrue" :newMess="newMess" :change:newMess="webRTC.changeNewMessage"
:isMedia="isMedia" :name="name" :change:name="webRTC.changeName" :change:isMedia="webRTC.changeIsMedia"
:animos="animos" :change:animos="changeAnimos" class="chat">
</view>3.连接顺序的问题，一定是：新进入的用户通过信令服务器给房间已有用户发送Offer，用户接收到Offer回应Answer，记住这个逻辑！
4.因为webRTC是运行在视图层的（也就是浏览器），而苹果默认浏览器是Safari，Safari浏览器默认机制是在用户主动和页面进行交互后，自动播放声音才会生效（也就是才有声音），所以在IOS端所有用户进入直播房间后默认都是静音的，用户主动开启音频才会受到直播间的声音（这是目前我发现的最好的解决办法）
五、核心代码（只有关键步骤）
1. 客户端socket
const socketTask = {
socket: null,
connect: () => {
getApp().globalData.socket = uni.connectSocket({
url:'ws://180.76.158.110:9000/socket/websocketv',
// url: 'ws://192.168.3.254:9000/socket/websocketv',
***plete: (e) => {
console.log(e);
},
});
getApp().globalData.socket.onOpen((data) => {
console.log("111111111");
getApp().globalData.socket.send({
data: JSON.stringify({
type: "newConnect",
userId: uni.getStorageSync('user').id,
})
})
})
getApp().globalData.socket.onClose((res) => {
console.log("连接关闭", res);
getApp().globalData.socket = null;
setTimeout(() => {
socketTask.connect()
}, 3000)
})
getApp().globalData.socket.onError((err) => {
console.log("连接异常", err);
getApp().globalData.socket = null;
setTimeout(() => {
socketTask.connect()
}, 1)
})
getApp().globalData.socket.onMessage((data) => {
socketTask.watchSocketMessage(data)
})
},
start: function() {
this.connect()
},
watchSocketMessage: function() {
// 这里实现自己的业务逻辑
}
}
export default socketTask2.客户端房间列表页
async onShow() {
if (!getApp().globalData.socket) {
await socketTask.start();
}
socketTask.watchSocketMessage = (data) => {
console.log("===========收到新消息==========",data);
this.watchSocketMessages(data)
}
},
methed:{
// 监听socket消息
watchSocketMessages(res) {
try {
const socket_msg = JSON.parse(res.data);
console.log("收到新消息", socket_msg);
switch (socket_msg.type) {
case "homeList":
if (socket_msg.data.length == 0) {
this.homeList = [];
uni.showToast({
title: "暂无房间，快去创建一个吧",
icon: "none"
})
} else {
this.homeList = socket_msg.data;
}
break
case "leave":
getApp().globalData.socket.send({
data: JSON.stringify({
type: "homeList",
userId: this.userInfo.userId,
})
})
break
case "createSu***ess":
uni.redirectTo({
url: `broadRoom?rid=${socket_msg.data.groupId}&&userId=${this.userInfo.id}&&groupInfo=${JSON.stringify(socket_msg.data)}`
})
break
}
} catch (e) {
}
},
}3.客户端直播间
逻辑层：
async onShow() {
const that = this;
if (!getApp().globalData.socket) {
console.log("socket不存在，重新连接");
await socketTask.start();
}
socketTask.watchSocketMessage = (data) => {
this.watchSocketMessage(data)
}
// 编译平台信息
uni.getSystemInfo({
su***ess(res) {
console.log("当前平台是", res);
if (res.osName == 'ios') {
console.log("我是ios", res)
that.isMedia = 'ios';
} else {
console.log("我是安卓", res)
that.isMedia = 'android';
}
}
})
}
methed：{
async watchSocketMessage(date) {
const data = JSON.parse(date.data);
switch (data.type) {
case "join":
console.log("join成功", data);
this.newMessaGes(data);
this.setUserList(data.admin);
this.updataNewMic(data)
// 找出自己以外的其他用户
const arr = this.userList.filter((item, index) => {
return item.userId !== this.userId
})
console.log("找出自己以外的其他用户", arr)
// 通知renderjs层创建RTC
this.RTCJoin = arr;
this.updataIsShow()
break
case "newjoin":
this.newMessaGes(data);
this.setUserList(data.admin);
break
case "offer":
//通知renderjs层有新人进入创建answer
console.log("收到offer", data)
this.RTCOffier = data;
break
case "answer":
// 找到对应peer,设置answer
console.log("收到offer", data)
this.RTCAnswer = data;
break
case "candidate":
// 找到对应的peer,将candidate添加进去
this.RT***andidate = data;
break
case "leave":
if (data.data == "房主已解散房间") {
this.closesAdmin()
} else {
const datas = {
data,
}
this.newMessaGes(datas)
this.setUserList(data.admin);
this.updataNewMic(data);
}
break
case "apply-admin":
this.updataIsApply(data.data)
break
case "newMic":
this.updataNewMic(data)
break
case "uplMicro":
this.updataNewMic(data)
break
case "newMessage":
this.newMess = data;
break
}
},
}视图层：
<script module="webRTC" lang="renderjs">
// 以下方法都在methed：{}中
// 监听changeRT***andidate
async changeRT***andidate(data) {
if (!data) {
return
}
console.log("this.otherPeerConnections", this.otherPeerConnections);
let arrs = this.otherPeerConnections.concat(this.myPeerConnections);
if (arrs.length == 0) {
return
}
let peerr = arrs.filter(item => {
return item.otherId == data.userId
})
if (peerr[0].peer == {}) {
return
} else {
console.log("candidatecandidate", data.candidate)
await peerr[0].peer.addIceCandidate(new RTCIceCandidate(data.candidate))
}
},
// 监听answer,找到对应peer设置answer
async changeRTCAnswer(data) {
if (!data) {
return
}
let peers = this.myPeerConnections.filter(item => {
return item.otherId == data.userId
})
console.log("peers[0]", peers[0])
await peers[0].peer.setRemoteDescription(new RTCSessionDescription(data.answer))
},
// 监听offier,RTCAnswer的创建
async changeRTCoffier(data) {
if (!data) {
return
}
let pear = null;
try {
pear = new RTCPeerConnection(iceServers);
} catch (e) {
console.log("实例化RTC-pear失败", e);
}
// 将音频流加入到Peer中
this.localStream.getAudioTracks()[0].enabled = this.isTrue;
this.localStream.getTracks().forEach(
(track) => pear.addTrack(track, this.localStream)
);
this.otherPeerConnections.push({
peer: pear,
otherId: data.userId
})
//当远程用户向对等连接添加流时，我们将显示它
pear.ontrack = (event) => {
// 为该用户创建audio
const track = event.track || event.streams[0]?.getTracks()[0];
if (track && track.kind === 'audio') {
console.log("存在音轨", event.streams[0]);
this.renderAudio(data.userId, event.streams[0]);
} else {
console.warn("No audio track found in the received stream.");
}
};
// 通过监听onicecandidate事件获取candidate信息
pear.onicecandidate = async (event) => {
if (event.candidate) {
// 通过信令服务器发送candidate信息给用户B
await this.$ownerInstance.callMethod("sendCandidate", {
type: "candidate",
userId: this.userId,
rid: this.rid,
msg: event.candidate,
formUserId: data.userId,
})
}
}
pear.setRemoteDescription(new RTCSessionDescription(data.offer))
// 接收端创建answer并发送给发起端
pear.createAnswer().then(answer => {
pear.setLocalDescription(answer);
// 通知serve层给房间用户发送answer
this.$ownerInstance.callMethod("sendAnswer", {
type: "answer",
userId: this.userId,
rid: this.rid,
msg: answer,
formUserId: data.userId,
})
})
},
// 发起连接申请,offier的创建
changeRTCjoin(RTCjoin) {
if (!RTCjoin) {
return
}
RTCjoin.forEach((item, index) => {
let peer = null;
try {
peer = new RTCPeerConnection(iceServers);
} catch (e) {
console.log("实例化RTC失败", e);
}
this.localStream.getAudioTracks()[0].enabled = this.isTrue;
this.localStream.getTracks().forEach(
(track) => peer.addTrack(track, this.localStream)
);
peer.ontrack = (event) => {
console.log("发起连接申请,offier的创建:peer.ontrack");
const track = event.track || event.streams[0]?.getTracks()[0];
if (track && track.kind === 'audio') {
console.log("存在音轨2", event.streams[0]);
this.renderAudio(item.userId, event.streams[0]);
} else {
console.warn("No audio track found in the received stream.");
}
};
// 通过监听onicecandidate事件获取candidate信息
peer.onicecandidate = (event) => {
if (event.candidate) {
// 通过信令服务器发送candidate信息给用户B
this.$ownerInstance.callMethod("sendCandidate", {
type: "candidate",
userId: this.userId,
rid: this.rid,
msg: event.candidate,
formUserId: item.userId,
})
}
}
this.myPeerConnections.push({
peer: peer,
otherId: item.userId
})
peer.createOffer(this.offerOptions).then(offer => {
peer.setLocalDescription(offer);
// 通知serve层给房间用户发送offier
this.$ownerInstance.callMethod("sendOffier", {
type: "offer",
userId: this.userId,
rid: this.rid,
msg: offer,
formUserId: item.userId,
})
})
})
},
renderAudio(uid, stream) {
let audio2 = document.getElementById(`audio_${uid}`);
console.log("audio_name", `audio_${uid}`);
if (!audio2) {
audio2 = document.createElement('audio');
audio2.id = `audio_${uid}`;
audio2.setAttribute("webkit-playsinline", "");
audio2.setAttribute("autoplay", true);
audio2.setAttribute("playsinline", "");
audio2.onloadedmetadata = () => {
if (this.isAudio == 1) {
console.log("不自动播放");
audio2.pause();
} else {
audio2.play();
}
};
this.audioList.push(audio2)
}
if ("srcObject" in audio2) {
console.log("使用了srcObject赋值");
audio2.srcObject = stream;
} else {
console.log("找不到srcObject赋值");
audio2.src = window.URL.createObjectURL(stream);
}
},
async initMedia() {
const that = this;
console.log("##########", this.isMedia);
// #ifdef APP-PLUS
if (this.isMedia == 'android') {
console.log("androidandroidandroidandroid");
await plus.android.requestPermissions(
['android.permission.RECORD_AUDIO'],
async (resultObj) => {
var result = 0;
for (var i = 0; i < resultObj.granted.length; i++) {
var grantedPermission = resultObj.granted[i];
result = 1
}
for (var i = 0; i < resultObj.deniedPresent.length; i++) {
var deniedPresentPermission = resultObj.deniedPresent[i];
result = 0
}
for (var i = 0; i < resultObj.deniedAlways.length; i++) {
var deniedAlwaysPermission = resultObj.deniedAlways[i];
result = -1
}
that.localStream = await that.getUserMedia();
that.$ownerInstance.callMethod("sendJoin", {
type: "join",
userId: that.userId,
rid: that.rid,
name: that.name
})
},
function(error) {
console.log("导入android出现错误", error);
}
);
} else {
console.log("iosiosiosiosiosios");
that.localStream = await that.getUserMedia().catch(err => {
console.log("出错了", err);
})
that.$ownerInstance.callMethod("sendJoin", {
type: "join",
userId: that.userId,
rid: that.rid,
name: that.name
})
}
// #endif
// #ifdef H5
that.localStream = await that.getUserMedia();
// 通知serve层加入成功
this.$ownerInstance.callMethod("sendJoin", {
type: "join",
userId: this.userId,
rid: this.rid,
name: this.name
})
// #endif
},
getUserMedia(then) {
return new Promise((resolve, reject) => {
navigator.mediaDevices.getUserMedia(this.mediaConstraints).then((stream) => {
return resolve(stream);
}).catch(err => {
if (err.name === 'NotAllowedError' || err.name === 'PermissionDeniedError') {
// 用户拒绝了授权
reject(new Error('用户拒绝了访问摄像头和麦克风的请求'));
} else if (err.name === 'NotFoundError' || err.name === 'DevicesNotFoundError') {
// 没有找到摄像头或麦克风
reject(new Error('没有找到摄像头或麦克风'));
} else if (err.name === 'NotReadableError' || err.name === 'TrackStartError') {
// 摄像头或麦克风不可读
reject(new Error('摄像头或麦克风不可读'));
} else if (err.name === 'OverconstrainedError' || err.name ===
'ConstraintNotSatisfiedError') {
// 由于媒体流的约束条件无法满足，请求被拒绝
reject(new Error('请求被拒绝，由于媒体流的约束条件无法满足'));
} else if (err.name === 'TypeError' || err.name === 'TypeError') {
// 发生了类型错误
reject(new Error('发生了类型错误'));
} else {
// 其他未知错误
reject(new Error('发生了未知错误'));
}
})
});
},
</script>4.信令服务器
略（就是socket，里面写swich，不会私信，小额收费）
