WebRTC音频通话升级为视频通话
我们有时候在音频通话过程中,想要改成视频通话。如果挂断当前通话再重新发起视频通话就会显得比较麻烦。
因此很多app提供了将音频通话升级成视频通话的功能,同时也有将视频通话降为音频通话的功能。
本文演示的是在本地模拟音频通话,并且将音频通话升级为视频通话。
准备
界面很简单,2个video加上几个按钮。
<video id="localVideo" playsinline autoplay muted></video>
<video id="remoteVideo" playsinline autoplay></video>
<div>
<button id="startBtn">开始</button>
<button id="callBtn">Call</button>
<button id="upgradeBtn">升级为视频通话</button>
<button id="hangupBtn">挂断</button>
</div>
用的是本地的adapter
<script src="../../src/js/adapter-2021.js"></script>
js
先来把元素拿到
const startBtn = document.getElementById('startBtn');
const callBtn = document.getElementById('callBtn');
const upgradeToVideoBtn = document.getElementById('upgradeBtn');
const hangupBtn = document.getElementById('hangupBtn');
const localVideo = document.getElementById('localVideo'); // 本地预览
const remoteVideo = document.getElementById('remoteVideo'); // 接收方
监听器
设置一些监听
localVideo.addEventListener('loadedmetadata', function () {
console.log(`localVideo 宽高: ${this.videoWidth}px, ${this.videoHeight}px`);
});
remoteVideo.addEventListener('loadedmetadata', function () {
console.log(`remoteVideo 宽高: ${this.videoWidth}px, ${this.videoHeight}px`);
});
let startTime;
remoteVideo.onresize = () => {
console.log(`remoteVideo onresize 宽高: ${remoteVideo.videoWidth}x${remoteVideo.videoHeight}`);
if (startTime) {
const elapsedTime = window.performance.now() - startTime;
console.log(`建立连接耗时: ${elapsedTime.toFixed(3)}ms`);
startTime = null;
}
};
startBtn.onclick = start;
callBtn.onclick = call;
upgradeToVideoBtn.onclick = upgrade;
hangupBtn.onclick = hangup;
打一些状态变化的log
function onCreateSessionDescriptionError(error) {
console.log(`rustfisher.com:创建会话描述失败, session description err: ${error.toString()}`);
}
function onIceStateChange(pc, event) {
if (pc) {
console.log(`rustfisher.com:${getName(pc)} ICE状态: ${pc.iceConnectionState}`);
console.log('rustfisher.com:ICE状态变化: ', event);
}
}
function onAddIceCandidateSuccess(pc) {
console.log(`rustfisher.com:${getName(pc)} addIceCandidate success 添加ICE候选成功`);
}
function onAddIceCandidateError(pc, error) {
console.log(`rustfisher.com:${getName(pc)} 添加ICE候选失败 failed to add ICE Candidate: ${error.toString()}`);
}
function onSetLocalSuccess(pc) {
console.log(`rustfisher.com:${getName(pc)} setLocalDescription 成功`);
}
function onSetSessionDescriptionError(error) {
console.log(`rustfisher.com:设置会话描述失败: ${error.toString()}`);
}
function onSetRemoteSuccess(pc) {
console.log(`rustfisher.com:${getName(pc)} 设置远程描述成功 setRemoteDescription complete`);
}
// 辅助方法
function getName(pc) {
return (pc === pc1) ? 'pc1' : 'pc2';
}
function getOtherPc(pc) {
return (pc === pc1) ? pc2 : pc1;
}
开始
获取本地的音频数据流,交给localVideo
function gotStream(stream) {
console.log('获取到了本地数据流');
localVideo.srcObject = stream;
localStream = stream;
callBtn.disabled = false;
}
function start() {
console.log('请求本地数据流 纯音频');
startBtn.disabled = true;
navigator.mediaDevices
.getUserMedia({ audio: true, video: false })
.then(gotStream)
.catch(e => alert(`getUserMedia() error: ${e.name}`));
}
call
发起音频呼叫
function call() {
callBtn.disabled = true;
upgradeToVideoBtn.disabled = false;
hangupBtn.disabled = false;
console.log('开始呼叫...');
startTime = window.performance.now();
const audioTracks = localStream.getAudioTracks();
if (audioTracks.length > 0) {
console.log(`使用的音频设备: ${audioTracks[0].label}`);
}
const servers = null; // 就在本地测试
pc1 = new RTCPeerConnection(servers);
console.log('创建本地节点 pc1');
pc1.onicecandidate = e => onIceCandidate(pc1, e);
pc2 = new RTCPeerConnection(servers);
console.log('rustfisher.com:创建模拟远端节点 pc2');
pc2.onicecandidate = e => onIceCandidate(pc2, e);
pc1.oniceconnectionstatechange = e => onIceStateChange(pc1, e);
pc2.oniceconnectionstatechange = e => onIceStateChange(pc2, e);
pc2.ontrack = gotRemoteStream;
localStream.getTracks().forEach(track => pc1.addTrack(track, localStream));
console.log('rustfisher.com:将本地数据流交给pc1');
console.log('rustfisher.com:pc1开始创建offer');
pc1.createOffer(offerOptions).then(onCreateOfferSuccess, onCreateSessionDescriptionError);
}
function gotRemoteStream(e) {
console.log('获取到远程数据流', e.track, e.streams[0]);
remoteVideo.srcObject = null;
remoteVideo.srcObject = e.streams[0];
}
function onIceCandidate(pc, event) {
getOtherPc(pc)
.addIceCandidate(event.candidate)
.then(() => onAddIceCandidateSuccess(pc), err => onAddIceCandidateError(pc, err));
console.log(`${getName(pc)} ICE candidate:\n${event.candidate ? event.candidate.candidate : '(null)'}`);
}
function onCreateOfferSuccess(desc) {
console.log(`pc1提供了offer\n${desc.sdp}`);
console.log('pc1 setLocalDescription start');
pc1.setLocalDescription(desc).then(() => onSetLocalSuccess(pc1), onSetSessionDescriptionError);
console.log('pc2 setRemoteDescription start');
pc2.setRemoteDescription(desc).then(() => onSetRemoteSuccess(pc2), onSetSessionDescriptionError);
console.log('pc2 createAnswer start');
pc2.createAnswer().then(onCreateAnswerSuccess, onCreateSessionDescriptionError);
}
function onCreateAnswerSuccess(desc) {
console.log(`rustfisher.com:pc2应答成功: ${desc.sdp}`);
console.log('pc2 setLocalDescription start');
pc2.setLocalDescription(desc).then(() => onSetLocalSuccess(pc2), onSetSessionDescriptionError);
console.log('pc1 setRemoteDescription start');
pc1.setRemoteDescription(desc).then(() => onSetRemoteSuccess(pc1), onSetSessionDescriptionError);
}
- 创建RTCPeerConnection
- 设置
onicecandidate
监听ICE候选 - 设置
oniceconnectionstatechange
监听ICE连接状态变化 - 接收方监听
ontrack
- 发送方pc1
addTrack
将当前数据流添加进去 - 发送方pc1创建offer
createOffer
- pc1创建好offer后,接收方pc2应答
createAnswer
升级到视频通话
upgrade()
方法处理升级操作
function upgrade() {
upgradeToVideoBtn.disabled = true;
navigator.mediaDevices
.getUserMedia({ video: true })
.then(stream => {
console.log('rustfisher.com:获取到了视频流');
const videoTracks = stream.getVideoTracks();
if (videoTracks.length > 0) {
console.log(`video device: ${videoTracks[0].label}`);
}
localStream.addTrack(videoTracks[0]);
localVideo.srcObject = null; // 重置视频流
localVideo.srcObject = localStream;
pc1.addTrack(videoTracks[0], localStream);
return pc1.createOffer();
})
.then(offer => pc1.setLocalDescription(offer))
.then(() => pc2.setRemoteDescription(pc1.localDescription))
.then(() => pc2.createAnswer())
.then(answer => pc2.setLocalDescription(answer))
.then(() => pc1.setRemoteDescription(pc2.localDescription));
}
发送方去获取音频数据流getUserMedia
。
将音频轨道添加进localStream
,并且发送方也要添加轨道 pc1.addTrack
。
创建offer createOffer
后面就是接收方pc2应答
挂断
简单的挂断功能如下
function hangup() {
console.log('rustfisher.com:挂断');
pc1.close();
pc2.close();
pc1 = null;
pc2 = null;
const videoTracks = localStream.getVideoTracks();
videoTracks.forEach(videoTrack => {
videoTrack.stop();
localStream.removeTrack(videoTrack);
});
localVideo.srcObject = null;
localVideo.srcObject = localStream;
hangupBtn.disabled = true;
callBtn.disabled = false;
}
主要是把呼出方的流关闭掉
代码流程描述图
将用户的操作(按钮)和主要代码对应起来
效果预览
效果预览请参考WebRTC音频通话升级到视频通话