如何基于uniapp开发android播放webrtc流详解

目录

• 一、播放rtsp协议流
• 二、播放webrtc协议流
• 三、音视频实时通讯
• 总结

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

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

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：

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
		......
		......
	}
}