WebRTC本地选择codec(web本地模拟)
codec:编码译码器,编解码器。它是一个程序,也可以是算法,或者设备,用于编码(encode)和解码(decode)数据流。
WebRTC能让两个web或者app之间建立音视频通信。通信过程中,数据流的格式必须被两边的设备支持。
WebRTC提供了接口查询支持的codec,并且可以设置要使用的codec。本文演示选择视频codec的过程。
示例
用户可以在发送视频流之前选择codec。把支持的codec类型列出来,用户自行选择。
开启视频后,建立连接前,我们可以选择设置codec。如上图蓝色区域所示。
html
先来准备页面。2个video控件分别显示收发视频。
按钮分别控制开始,呼叫(发起连接)和挂断。
select
用来选择codec。获取支持的codec信息,放到下拉栏里让用户选择。
以下是index.html主要内容
<div id="container">
<h1><a href="https://an.rustfisher.com/webrtc/peerconnection/change-codec/" title="WebRTC示例,修改codec">WebRTC示例,修改codec</a>
</h1>
<video id="localVideo" playsinline autoplay muted></video>
<video id="remoteVideo" playsinline autoplay></video>
<div class="box">
<button id="startBtn">开始</button>
<button id="callBtn">呼叫</button>
<button id="hangupBtn">挂断</button>
</div>
<div class="box">
<span>选择Codec:</span>
<select id="codecPreferences" disabled>
<option selected value="">Default</option>
</select>
<div id="actualCodec"></div>
</div>
<p>可以在控制台观察 <code>MediaStream</code>, <code>localStream</code>, 和 <code>RTCPeerConnection</code></p>
</div>
<script src="../../src/js/adapter-2021.js"></script>
<script src="js/main.js" async></script>
adapter-2021.js是存放在本地的文件。要使用最新的adapter,按以下地址引入
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
js
main.js控制主要逻辑。从开启摄像头开始。建立连接前可以选择codec。
建立连接的流程与「WebRTC模拟传输视频流,video通过本地节点peer传输视频流」类似。
获取可用codec
先判断浏览器是否有RTCRtpTransceiver
,并且要能支持setCodecPreferences
方法
const supportsSetCodecPreferences = window.RTCRtpTransceiver &&
'setCodecPreferences' in window.RTCRtpTransceiver.prototype;
通过RTCRtpSender.getCapabilities('video')
获取可支持的codec。
然后把它们放进列表codecPreferences
里
if (supportsSetCodecPreferences) {
const { codecs } = RTCRtpSender.getCapabilities('video');
codecs.forEach(codec => {
if (['video/red', 'video/ulpfec', 'video/rtx'].includes(codec.mimeType)) {
return;
}
const option = document.createElement('option');
option.value = (codec.mimeType + ' ' + (codec.sdpFmtpLine || '')).trim();
option.innerText = option.value;
codecPreferences.appendChild(option);
});
codecPreferences.disabled = false;
}
配置codec
呼叫之前,找到用户选择的codec。
调用transceiver.setCodecPreferences(codecs)
,把选中的codec交给transceiver
。
if (supportsSetCodecPreferences) {
// 获取选择的codec
const preferredCodec = codecPreferences.options[codecPreferences.selectedIndex];
if (preferredCodec.value !== '') {
const [mimeType, sdpFmtpLine] = preferredCodec.value.split(' ');
const { codecs } = RTCRtpSender.getCapabilities('video');
const selectedCodecIndex = codecs.findIndex(c => c.mimeType === mimeType && c.sdpFmtpLine === sdpFmtpLine);
const selectedCodec = codecs[selectedCodecIndex];
codecs.splice(selectedCodecIndex, 1);
codecs.unshift(selectedCodec);
console.log(codecs);
const transceiver = pc1.getTransceivers().find(t => t.sender && t.sender.track === localStream.getVideoTracks()[0]);
transceiver.setCodecPreferences(codecs);
console.log('选择的codec', selectedCodec);
}
}
main.js完整代码如下
'use strict';
console.log('WebRTC示例,选择codec');
// --------- ui准备 ---------
const startBtn = document.getElementById('startBtn');
const callBtn = document.getElementById('callBtn');
const hangupBtn = document.getElementById('hangupBtn');
const localVideo = document.getElementById('localVideo');
const remoteVideo = document.getElementById('remoteVideo');
callBtn.disabled = true;
hangupBtn.disabled = true;
startBtn.addEventListener('click', start);
callBtn.addEventListener('click', call);
hangupBtn.addEventListener('click', hangup);
// ---------------------------
// -------- codec 的配置 --------
const codecPreferences = document.querySelector('#codecPreferences');
const supportsSetCodecPreferences = window.RTCRtpTransceiver &&
'setCodecPreferences' in window.RTCRtpTransceiver.prototype;
// -----------------------------
let startTime;
remoteVideo.addEventListener('resize', () => {
console.log(`Remote video size changed to ${remoteVideo.videoWidth}x${remoteVideo.videoHeight}`);
if (startTime) {
const elapsedTime = window.performance.now() - startTime;
console.log('视频流连接耗时: ' + elapsedTime.toFixed(3) + 'ms');
startTime = null;
}
});
let localStream;
let pc1;
let pc2;
const offerOptions = {
offerToReceiveAudio: 1,
offerToReceiveVideo: 1
};
function getName(pc) {
return (pc === pc1) ? 'pc1' : 'pc2';
}
function getOtherPc(pc) {
return (pc === pc1) ? pc2 : pc1;
}
// 启动本地视频
async function start() {
console.log('启动本地视频');
startBtn.disabled = true;
try {
const stream = await navigator.mediaDevices.getUserMedia({ audio: true, video: true });
console.log('获取到本地视频');
localVideo.srcObject = stream;
localStream = stream;
callBtn.disabled = false;
} catch (e) {
alert(`getUserMedia() error: ${e.name}`);
}
if (supportsSetCodecPreferences) {
const { codecs } = RTCRtpSender.getCapabilities('video');
console.log('RTCRtpSender.getCapabilities(video):\n', codecs);
codecs.forEach(codec => {
if (['video/red', 'video/ulpfec', 'video/rtx'].includes(codec.mimeType)) {
return;
}
const option = document.createElement('option');
option.value = (codec.mimeType + ' ' + (codec.sdpFmtpLine || '')).trim();
option.innerText = option.value;
codecPreferences.appendChild(option);
});
codecPreferences.disabled = false;
} else {
console.warn('当前不支持更换codec');
}
}
// 呼叫并建立连接
async function call() {
callBtn.disabled = true;
hangupBtn.disabled = false;
console.log('开始呼叫');
startTime = window.performance.now();
const videoTracks = localStream.getVideoTracks();
const audioTracks = localStream.getAudioTracks();
if (videoTracks.length > 0) {
console.log(`使用的摄像头: ${videoTracks[0].label}`);
}
if (audioTracks.length > 0) {
console.log(`使用的麦克风: ${audioTracks[0].label}`);
}
const configuration = {};
pc1 = new RTCPeerConnection(configuration);
pc1.addEventListener('icecandidate', e => onIceCandidate(pc1, e));
pc2 = new RTCPeerConnection(configuration);
pc2.addEventListener('icecandidate', e => onIceCandidate(pc2, e));
pc2.addEventListener('track', gotRemoteStream);
localStream.getTracks().forEach(track => pc1.addTrack(track, localStream));
if (supportsSetCodecPreferences) {
// 获取选择的codec
const preferredCodec = codecPreferences.options[codecPreferences.selectedIndex];
if (preferredCodec.value !== '') {
const [mimeType, sdpFmtpLine] = preferredCodec.value.split(' ');
const { codecs } = RTCRtpSender.getCapabilities('video');
const selectedCodecIndex = codecs.findIndex(c => c.mimeType === mimeType && c.sdpFmtpLine === sdpFmtpLine);
const selectedCodec = codecs[selectedCodecIndex];
codecs.splice(selectedCodecIndex, 1);
codecs.unshift(selectedCodec);
console.log(codecs);
const transceiver = pc1.getTransceivers().find(t => t.sender && t.sender.track === localStream.getVideoTracks()[0]);
transceiver.setCodecPreferences(codecs);
console.log('选择的codec', selectedCodec);
}
}
codecPreferences.disabled = true;
try {
const offer = await pc1.createOffer(offerOptions);
await onCreateOfferSuccess(offer);
} catch (e) {
console.log(`Failed, pc1 createOffer: ${e.toString()}`);
}
}
async function onCreateOfferSuccess(desc) {
try {
await pc1.setLocalDescription(desc);
console.log('pc1 setLocalDescription 成功');
} catch (e) {
console.error('pc1 setLocalDescription 出错', e);
}
try {
await pc2.setRemoteDescription(desc);
console.log('pc2 setRemoteDescription ok');
} catch (e) {
console.error('pc2 setRemoteDescription fail', e);
}
try {
const answer = await pc2.createAnswer();
await onCreateAnswerSuccess(answer);
} catch (e) {
console.log(`pc2 create answer fail: ${e.toString()}`);
}
}
function gotRemoteStream(e) {
if (remoteVideo.srcObject !== e.streams[0]) {
remoteVideo.srcObject = e.streams[0];
console.log('pc2 received remote stream');
}
}
// 应答(接收)成功
async function onCreateAnswerSuccess(desc) {
console.log(`Answer from pc2:\n${desc.sdp}`);
console.log('pc2 setLocalDescription start');
try {
await pc2.setLocalDescription(desc);
} catch (e) {
console.error('pc2 set local d fail', e);
}
console.log('pc1 setRemoteDescription start');
try {
await pc1.setRemoteDescription(desc);
// Display the video codec that is actually used.
setTimeout(async () => {
const stats = await pc1.getStats();
stats.forEach(stat => {
if (!(stat.type === 'outbound-rtp' && stat.kind === 'video')) {
return;
}
const codec = stats.get(stat.codecId);
document.getElementById('actualCodec').innerText = 'Using ' + codec.mimeType +
' ' + (codec.sdpFmtpLine ? codec.sdpFmtpLine + ' ' : '') +
', payloadType=' + codec.payloadType + '. Encoder: ' + stat.encoderImplementation;
});
}, 1000);
} catch (e) {
console.error(e);
}
}
async function onIceCandidate(pc, event) {
try {
await (getOtherPc(pc).addIceCandidate(event.candidate));
onAddIceCandidateSuccess(pc);
} catch (e) {
onAddIceCandidateError(pc, e);
}
console.log(`${getName(pc)} ICE candidate:\n${event.candidate ? event.candidate.candidate : '(null)'}`);
}
function onAddIceCandidateSuccess(pc) {
console.log(`${getName(pc)} addIceCandidate success`);
}
function onAddIceCandidateError(pc, error) {
console.log(`${getName(pc)} failed to add ICE Candidate: ${error.toString()}`);
}
localVideo.addEventListener('loadedmetadata', function () {
console.log(`Local video videoWidth: ${this.videoWidth}px, videoHeight: ${this.videoHeight}px`);
});
remoteVideo.addEventListener('loadedmetadata', function () {
console.log(`Remote video videoWidth: ${this.videoWidth}px, videoHeight: ${this.videoHeight}px`);
});
// 挂断
function hangup() {
console.log('挂断');
pc1.close();
pc2.close();
pc1 = null;
pc2 = null;
hangupBtn.disabled = true;
callBtn.disabled = false;
codecPreferences.disabled = false;
}
codec信息说明
观察控制台,打印出了可用codec信息(Mac,97.0.4692.71(正式版本)x86_64)。主要关注下面3种
{clockRate: 90000, mimeType: 'video/VP8'}
{clockRate: 90000, mimeType: 'video/VP9', sdpFmtpLine: 'profile-id=0'}
{clockRate: 90000, mimeType: 'video/H264', sdpFmtpLine: 'level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f'}
clockRate
是codec的时钟频率,单位hz
sdpFmtpLine
是codec的SDP里a=fmtp
的参数信息
mimeType
里说的是视频编码类型,常见的有VP8和H264等等
支持WebRTC的浏览器,必须要支持视频codec VP8和H264
VP8与VP9
2010年5月Google收购了On2 Technologies,获得了VP8。
Opera,FireFox,Chrome和Chromium支持HTML5中的video播放VP8视频。
WebM作为一个容器格式,图像部分使用VP8,音频使用Vorbis和Opus。
VP9由Google开发,一个开放的无版权费的视频编码标准。开发初期曾用名“Next Gen Open Video”。VP9也被视为是VP8的下一代视频编码标准。
H264
H.264,又称为MPEG-4第10部分,高级视频编码是一种面向块,基于运动补偿的视频编码标准。
到2014年,它已经成为高精度视频录制、压缩和发布的最常用格式之一。
优势:
- 1)网络亲和性,即可适用于各种传输网络
- 2)高的视频压缩比
目前我们用的比较多的还是H264。
效果预览
网页效果请参考 选择codec