如何基于uniapp开发android播放webrtc流详解
这篇文章主要介绍了在uniapp中播放RTSP和WebRTC协议流的区别,以及如何封装WebrtcVideo组件和音视频实时通讯的实现,文中通过代码介绍的非常详细,需要的朋友可以参考下
一、播放rtsp协议流
如果 webrtc 流以 rtsp 协议返回，流地址如：rtsp://127.0.0.1:5115/session.mpg，uniapp的 编译到android上直接就能播放，但通常会有2-3秒的延迟。
二、播放webrtc协议流
如果 webrtc 流以 webrtc 协议返回，流地址如：webrtc://127.0.0.1:1988/live/livestream，我们需要通过sdp协商、连接推流服务端、搭建音视频流通道来播放音视频流，通常有500毫秒左右的延迟。
封装 WebrtcVideo 组件
<template>
<video id="rtc_media_player" width="100%" height="100%" autoplay playsinline></video>
</template>
<!-- 因为我们使用到 js 库，所以需要使用 uniapp 的 renderjs -->
<script module="webrtcVideo" lang="renderjs">
import $ from "./jquery-1.10.2.min.js";
import {prepareUrl} from "./utils.js";
export default {
data() {
return {
//RTCPeerConnection 对象
peerConnection: null,
//需要播放的webrtc流地址
playUrl: 'webrtc://127.0.0.1:1988/live/livestream'
}
},
methods: {
createPeerConnection() {
const that = this
//创建 WebRTC 通信通道
that.peerConnection = new RTCPeerConnection(null);
//添加一个单向的音视频流收发器
that.peerConnection.addTransceiver("audio", { direction: "recvonly" });
that.peerConnection.addTransceiver("video", { direction: "recvonly" });
//收到服务器码流，将音视频流写入播放器
that.peerConnection.ontrack = (event) => {
const remoteVideo = document.getElementById("rtc_media_player");
if (remoteVideo.srcObject !== event.streams[0]) {
remoteVideo.srcObject = event.streams[0];
}
};
},
async makeCall() {
const that = this
const url = this.playUrl
this.createPeerConnection()
//拼接服务端请求地址，如：http://192.168.0.1:1988/rtc/v1/play/
const conf = prepareUrl(url);
//生成 offer sdp
const offer = await this.peerConnection.createOffer();
await this.peerConnection.setLocalDescription(offer);
var session = await new Promise(function (resolve, reject) {
$.ajax({
type: "POST",
url: conf.apiUrl,
data: offer.sdp,
contentType: "text/plain",
dataType: "json",
crossDomain: true,
})
.done(function (data) {
//服务端返回 answer sdp
if (data.code) {
reject(data);
return;
}
resolve(data);
})
.fail(function (reason) {
reject(reason);
});
});
//设置远端的描述信息，协商sdp，通过后搭建通道成功
await this.peerConnection.setRemoteDescription(
new RTCSessionDescription({ type: "answer", sdp: session.sdp })
);
session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'
return session;
}
},
mounted() {
try {
this.makeCall().then((res) => {
// webrtc 通道建立成功
})
} catch (error) {
// webrtc 通道建立失败
console.log(error)
}
}
}
</script>
utils.js
const defaultPath = "/rtc/v1/play/";
export const prepareUrl = webrtcUrl => {
var urlObject = parseUrl(webrtcUrl);
var schema = "http:";
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 || defaultPath;
if (api.lastIndexOf("/") !== api.length - 1) {
api += "/";
}
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
var 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)
.substr(0, 7)
};
};
export const parseUrl = 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.substr(1, a.pathname.lastIndexOf("/") - 1);
var stream = a.pathname.substr(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.substr(app.indexOf("?"));
app = app.substr(0, app.indexOf("?"));
if (params.indexOf("vhost=") > 0) {
vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
if (vhost.indexOf("&") > 0) {
vhost = vhost.substr(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.substr(0, url.indexOf("://"));
}
var port = a.port;
if (!port) {
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
};
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;
};
export const fill_query = (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;
}
};
页面中使用
<template>
<VideoWebrtc />
</template>
<script setup>
import VideoWebrtc from "@/components/videoWebrtc";
</script>
需要注意的事项：
1.spd 协商的重要标识之一为媒体描述： m=xxx ，示例行如下：
一个完整的媒体描述，从第一个m=xxx 开始，到下一个m=xxx 结束，以video为例，媒体描述包含了当前设备允许播放的视频流编码格式，常见如：VP8/VP9/H264 等：
对照 m=video 后边的编码发现，其包含所有 a=rtpmap 后的编码，a=rtpmap 编码后的字符串代表视频流格式，但视频编码与视频流格式却不是固定的匹配关系，也就是说，在设备A中，可能存在 a=rtpmap:106 H264/90000 表示h264，在设备B中，a=rtpmap:100 H264/90000 表示h264。
因此，如果要鉴别设备允许播放的视频流格式，我们需要观察 a=rtpmap code 后的字符串。
协商通过的部分标准为：
offer sdp 的 m=xxx 数量需要与 answer sdp 的 m=xxx 数量保持一致；
offer sdp 的 m=xxx 顺序需要与 answer sdp 的 m=xxx 顺序保持一致；如两者都需要将 m=audio 放在第一位，m=video放在第二位，或者反过来；
answer sdp 返回的 m=audio 后的 ，需要被包含在 offer sdp 的 m=audio 后的中；
offer sdp 的 m=xxx 由 addTransceiver 创建，首个参数为 audio 时，生成 m=audio，首个参数为video时，生成 m=video ，创建顺序对应 m=xxx 顺序
1
2
"recvonly" }); that.peerConnection.addTransceiver("video", {
direction: "recvonly" }); ```
在 sdp 中存在一项 a=mid:xxx xxx在浏览器中可能为 audio、video ，在 android 设备上为 0、1，服务端需注意与 offer sdp 匹配。
关于音视频流收发器，上面使用的api是 addTransceiver ，但在部分android设备上会提示没有这个api，我们可以替换为 getUserMedia + addTrack：
data() {
return {
......
localStream: null,
......
}
},
methods: {
createPeerConnection() {
const that = this
//创建 WebRTC 通信通道
that.peerConnection = new RTCPeerConnection(null);
that.localStream.getTracks().forEach((track) => {
that.peerConnection.addTrack(track, that.localStream);
});
//收到服务器码流，将音视频流写入播放器
that.peerConnection.ontrack = (event) => {
......
};
}，
async makeCall() {
const that = this
that.localStream = await navigator.mediaDevices.getUserMedia({
video: true,
audio: true,
});
const url = this.playUrl
......
......
}
}
需要注意的是，navigator.mediaDevices.getUserMedia获取的是设备摄像头、录音的媒体流，所以设备首先要具备摄像、录音功能，并开启对应权限，否则 api 将调用失败。
三、音视频实时通讯
这种 p2p 场景的流播放，通常需要使用 websocket 建立服务器连接，然后同时播放本地、服务端的流。
<template>
<div>Local Video</div>
<video id="localVideo" autoplay playsinline></video>
<div>Remote Video</div>
<video id="remoteVideo" autoplay playsinline></video>
</template>
<script module="webrtcVideo" lang="renderjs">
import $ from "./jquery-1.10.2.min.js";
export default {
data() {
return {
signalingServerUrl: "ws://127.0.0.1:8085",
iceServersUrl: 'stun:stun.l.google.com:19302',
localStream: null,
peerConnection: null
}
},
methods: {
async startLocalStream(){
try {
this.localStream = await navigator.mediaDevices.getUserMedia({
video: true,
audio: true,
});
document.getElementById("localVideo").srcObject = this.localStream;
}catch (err) {
console.error("Error accessing media devices.", err);
}
},
createPeerConnection() {
const configuration = { iceServers: [{
urls: this.iceServersUrl
}]};
this.peerConnection = new RTCPeerConnection(configuration);
this.localStream.getTracks().forEach((track) => {
this.peerConnection.addTrack(track, this.localStream);
});
this.peerConnection.onicecandidate = (event) => {
if (event.candidate) {
ws.send(
JSON.stringify({
type: "candidate",
candidate: event.candidate,
})
);
}
};
this.peerConnection.ontrack = (event) => {
const remoteVideo = document.getElementById("remoteVideo");
if (remoteVideo.srcObject !== event.streams[0]) {
remoteVideo.srcObject = event.streams[0];
}
};
}，
async makeCall() {
this.createPeerConnection();
const offer = await this.peerConnection.createOffer();
await this.peerConnection.setLocalDescription(offer);
ws.send(JSON.stringify(offer));
}
},
mounted() {
this.makeCall()
const ws = new WebSocket(this.signalingServerUrl);
ws.onopen = () => {
console.log("Connected to the signaling server");
this.startLocalStream();
};
ws.onmessage = async (message) => {
const data = JSON.parse(message.data);
if (data.type === "offer") {
if (!this.peerConnection) createPeerConnection();
await this.peerConnection.setRemoteDescription(
new RTCSessionDescription(data)
);
const answer = await this.peerConnection.createAnswer();
await this.peerConnection.setLocalDescription(answer);
ws.send(JSON.stringify(this.peerConnection.localDescription));
} else if (data.type === "answer") {
if (!this.peerConnection) createPeerConnection();
await this.peerConnection.setRemoteDescription(
new RTCSessionDescription(data)
);
} else if (data.type === "candidate") {
if (this.peerConnection) {
try {
await this.peerConnection.addIceCandidate(
new RTCIceCandidate(data.candidate)
);
} catch (e) {
console.error("Error adding received ICE candidate", e);
}
}
}
}
}
}
</script>
播放webrtc协议流相比，p2p 以 WebSocket 替代 ajax 实现 sdp 的发送与接收，增加了本地流的播放功能，其他与播放协议流的代码一致。
总结
到此这篇关于如何基于uniapp开发android播放webrtc流的文章就介绍到这了