//
// Copyright (c) 2013-2021 Winlin
//
// SPDX-License-Identifier: MIT
//

"use strict";

function SrsError(name, message) {
	this.name = name;
	this.message = message;
	this.stack = new Error().stack;
}
SrsError.prototype = Object.create(Error.prototype);
SrsError.prototype.constructor = SrsError;

// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-awat-prmise based SRS RTC Publisher.
function SrsRtcPublisherAsync() {
	var self = {};

	// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
	self.constraints = {
		audio: true,
		video: {
			width: { ideal: 320, max: 576 }
		}
	};

	// @see https://github.com/rtcdn/rtcdn-draft
	// @url The WebRTC url to play with, for example:
	//      webrtc://r.ossrs.net/live/livestream
	// or specifies the API port:
	//      webrtc://r.ossrs.net:11985/live/livestream
	// or autostart the publish:
	//      webrtc://r.ossrs.net/live/livestream?autostart=true
	// or change the app from live to myapp:
	//      webrtc://r.ossrs.net:11985/myapp/livestream
	// or change the stream from livestream to mystream:
	//      webrtc://r.ossrs.net:11985/live/mystream
	// or set the api server to myapi.domain.com:
	//      webrtc://myapi.domain.com/live/livestream
	// or set the candidate(eip) of answer:
	//      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
	// or force to access https API:
	//      webrtc://r.ossrs.net/live/livestream?schema=https
	// or use plaintext, without SRTP:
	//      webrtc://r.ossrs.net/live/livestream?encrypt=false
	// or any other information, will pass-by in the query:
	//      webrtc://r.ossrs.net/live/livestream?vhost=xxx
	//      webrtc://r.ossrs.net/live/livestream?token=xxx
	self.publish = async function (url) {
		var conf = self.__internal.prepareUrl(url);
		self.pc.addTransceiver("audio", { direction: "sendonly" });
		self.pc.addTransceiver("video", { direction: "sendonly" });
		//self.pc.addTransceiver("video", {direction: "sendonly"});
		//self.pc.addTransceiver("audio", {direction: "sendonly"});

		if (!navigator.mediaDevices && window.location.protocol === "http:" && window.location.hostname !== "localhost") {
			throw new SrsError(
				"HttpsRequiredError",
				`Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`
			);
		}
		var stream = await navigator.mediaDevices.getUserMedia(self.constraints);

		// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
		stream.getTracks().forEach(function (track) {
			self.pc.addTrack(track);

			// Notify about local track when stream is ok.
			self.ontrack && self.ontrack({ track: track });
		});

		var offer = await self.pc.createOffer();
		await self.pc.setLocalDescription(offer);
		var session = await new Promise(function (resolve, reject) {
			// @see https://github.com/rtcdn/rtcdn-draft
			var data = {
				api: conf.apiUrl,
				tid: conf.tid,
				streamurl: conf.streamUrl,
				clientip: null,
				sdp: offer.sdp
			};
			console.log("Generated offer: ", data);

			const xhr = new XMLHttpRequest();
			xhr.onload = function () {
				if (xhr.readyState !== xhr.DONE) return;
				if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
				const data = JSON.parse(xhr.responseText);
				console.log("Got answer: ", data);
				return data.code ? reject(xhr) : resolve(data);
			};
			xhr.open("POST", conf.apiUrl, true);
			xhr.setRequestHeader("Content-type", "application/json");
			xhr.send(JSON.stringify(data));
		});
		await self.pc.setRemoteDescription(new RTCSessionDescription({ type: "answer", sdp: session.sdp }));
		session.simulator = conf.schema + "//" + conf.urlObject.server + ":" + conf.port + "/rtc/v1/nack/";

		return session;
	};

	// Close the publisher.
	self.close = function () {
		self.pc && self.pc.close();
		self.pc = null;
	};

	// The callback when got local stream.
	// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
	self.ontrack = function (event) {
		// Add track to stream of SDK.
		self.stream.addTrack(event.track);
	};

	// Internal APIs.
	self.__internal = {
		defaultPath: "/rtc/v1/publish/",
		prepareUrl: function (webrtcUrl) {
			var urlObject = self.__internal.parse(webrtcUrl);

			// If user specifies the schema, use it as API schema.
			var schema = urlObject.user_query.schema;
			schema = schema ? schema + ":" : window.location.protocol;

			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 || self.__internal.defaultPath;
			if (api.lastIndexOf("/") !== api.length - 1) {
				api += "/";
			}

			var 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
			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)
					.slice(0, 7)
			};
		},
		parse: function (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.substring(1, a.pathname.lastIndexOf("/"));
			var stream = a.pathname.slice(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.slice(app.indexOf("?"));
				app = app.slice(0, app.indexOf("?"));

				if (params.indexOf("vhost=") > 0) {
					vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
					if (vhost.indexOf("&") > 0) {
						vhost = vhost.slice(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.slice(0, url.indexOf("://"));
			}

			var port = a.port;
			if (!port) {
				// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
				if (schema === "webrtc" && url.indexOf(`webrtc://${a.host}:`) === 0) {
					port = url.indexOf(`webrtc://${a.host}:80`) === 0 ? 80 : 443;
				}

				// Guess by schema.
				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
			};
			self.__internal.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;
		},
		fill_query: function (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;
			}
		}
	};

	self.pc = new RTCPeerConnection(null);

	// To keep api consistent between player and publisher.
	// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
	// @see https://webrtc.org/getting-started/media-devices
	self.stream = new MediaStream();

	return self;
}

// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-await-promise based SRS RTC Player.
function SrsRtcPlayerAsync() {
	var self = {};

	// @see https://github.com/rtcdn/rtcdn-draft
	// @url The WebRTC url to play with, for example:
	//      webrtc://r.ossrs.net/live/livestream
	// or specifies the API port:
	//      webrtc://r.ossrs.net:11985/live/livestream
	//      webrtc://r.ossrs.net:80/live/livestream
	// or autostart the play:
	//      webrtc://r.ossrs.net/live/livestream?autostart=true
	// or change the app from live to myapp:
	//      webrtc://r.ossrs.net:11985/myapp/livestream
	// or change the stream from livestream to mystream:
	//      webrtc://r.ossrs.net:11985/live/mystream
	// or set the api server to myapi.domain.com:
	//      webrtc://myapi.domain.com/live/livestream
	// or set the candidate(eip) of answer:
	//      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
	// or force to access https API:
	//      webrtc://r.ossrs.net/live/livestream?schema=https
	// or use plaintext, without SRTP:
	//      webrtc://r.ossrs.net/live/livestream?encrypt=false
	// or any other information, will pass-by in the query:
	//      webrtc://r.ossrs.net/live/livestream?vhost=xxx
	//      webrtc://r.ossrs.net/live/livestream?token=xxx
	self.play = async function (url) {
		var conf = self.__internal.prepareUrl(url);
		self.pc.addTransceiver("audio", { direction: "recvonly" });
		self.pc.addTransceiver("video", { direction: "recvonly" });
		//self.pc.addTransceiver("video", {direction: "recvonly"});
		//self.pc.addTransceiver("audio", {direction: "recvonly"});

		var offer = await self.pc.createOffer();
		await self.pc.setLocalDescription(offer);
		var session = await new Promise(function (resolve, reject) {
			// @see https://github.com/rtcdn/rtcdn-draft
			var data = {
				api: conf.apiUrl,
				tid: conf.tid,
				streamurl: conf.streamUrl,
				clientip: null,
				sdp: offer.sdp
			};
			console.log("Generated offer: ", data);

			const xhr = new XMLHttpRequest();
			xhr.onload = function () {
				if (xhr.readyState !== xhr.DONE) return;
				if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
				const data = JSON.parse(xhr.responseText);
				console.log("Got answer: ", data);
				return data.code ? reject(xhr) : resolve(data);
			};
			xhr.open("POST", conf.apiUrl, true);
			xhr.setRequestHeader("Content-type", "application/json");
			xhr.send(JSON.stringify(data));
		});
		await self.pc.setRemoteDescription(new RTCSessionDescription({ type: "answer", sdp: session.sdp }));
		session.simulator = conf.schema + "//" + conf.urlObject.server + ":" + conf.port + "/rtc/v1/nack/";

		return session;
	};

	// Close the player.
	self.close = function () {
		self.pc && self.pc.close();
		self.pc = null;
	};

	// The callback when got remote track.
	// Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
	self.ontrack = function (event) {
		// https://webrtc.org/getting-started/remote-streams
		self.stream.addTrack(event.track);
	};

	// Internal APIs.
	self.__internal = {
		defaultPath: "/rtc/v1/play/",
		prepareUrl: function (webrtcUrl) {
			var urlObject = self.__internal.parse(webrtcUrl);

			// If user specifies the schema, use it as API schema.
			var schema = urlObject.user_query.schema;
			schema = schema ? schema + ":" : window.location.protocol;

			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 || self.__internal.defaultPath;
			if (api.lastIndexOf("/") !== api.length - 1) {
				api += "/";
			}

			var 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
			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)
					.slice(0, 7)
			};
		},
		parse: function (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.substring(1, a.pathname.lastIndexOf("/"));
			var stream = a.pathname.slice(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.slice(app.indexOf("?"));
				app = app.slice(0, app.indexOf("?"));

				if (params.indexOf("vhost=") > 0) {
					vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
					if (vhost.indexOf("&") > 0) {
						vhost = vhost.slice(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.slice(0, url.indexOf("://"));
			}

			var port = a.port;
			if (!port) {
				// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
				if (schema === "webrtc" && url.indexOf(`webrtc://${a.host}:`) === 0) {
					port = url.indexOf(`webrtc://${a.host}:80`) === 0 ? 80 : 443;
				}

				// Guess by schema.
				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
			};
			self.__internal.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;
		},
		fill_query: function (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;
			}
		}
	};

	self.pc = new RTCPeerConnection(null);

	// Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
	self.stream = new MediaStream();

	// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
	self.pc.ontrack = function (event) {
		if (self.ontrack) {
			self.ontrack(event);
		}
	};

	return self;
}

// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-awat-prmise based SRS RTC Publisher by WHIP.
function SrsRtcWhipWhepAsync() {
	var self = {};

	// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
	self.constraints = {
		audio: true,
		video: {
			width: { ideal: 320, max: 576 }
		}
	};

	// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
	// @url The WebRTC url to publish with, for example:
	//      http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
	self.publish = async function (url) {
		if (url.indexOf("/whip/") === -1) throw new Error(`invalid WHIP url ${url}`);

		self.pc.addTransceiver("audio", { direction: "sendonly" });
		self.pc.addTransceiver("video", { direction: "sendonly" });

		if (!navigator.mediaDevices && window.location.protocol === "http:" && window.location.hostname !== "localhost") {
			throw new SrsError(
				"HttpsRequiredError",
				`Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`
			);
		}
		var stream = await navigator.mediaDevices.getUserMedia(self.constraints);

		// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
		stream.getTracks().forEach(function (track) {
			self.pc.addTrack(track);

			// Notify about local track when stream is ok.
			self.ontrack && self.ontrack({ track: track });
		});

		var offer = await self.pc.createOffer();
		await self.pc.setLocalDescription(offer);
		const answer = await new Promise(function (resolve, reject) {
			console.log("Generated offer: ", offer);

			const xhr = new XMLHttpRequest();
			xhr.onload = function () {
				if (xhr.readyState !== xhr.DONE) return;
				if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
				const data = xhr.responseText;
				console.log("Got answer: ", data);
				return data.code ? reject(xhr) : resolve(data);
			};
			xhr.open("POST", url, true);
			xhr.setRequestHeader("Content-type", "application/sdp");
			xhr.send(offer.sdp);
		});
		await self.pc.setRemoteDescription(new RTCSessionDescription({ type: "answer", sdp: answer }));

		return self.__internal.parseId(url, offer.sdp, answer);
	};

	// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
	// @url The WebRTC url to play with, for example:
	//      http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
	self.play = async function (url) {
		if (url.indexOf("/whip-play/") === -1 && url.indexOf("/whep/") === -1) throw new Error(`invalid WHEP url ${url}`);

		self.pc.addTransceiver("audio", { direction: "recvonly" });
		self.pc.addTransceiver("video", { direction: "recvonly" });

		var offer = await self.pc.createOffer();
		await self.pc.setLocalDescription(offer);
		const answer = await new Promise(function (resolve, reject) {
			console.log("Generated offer: ", offer);

			const xhr = new XMLHttpRequest();
			xhr.onload = function () {
				if (xhr.readyState !== xhr.DONE) return;
				if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
				const data = xhr.responseText;
				console.log("Got answer: ", data);
				return data.code ? reject(xhr) : resolve(data);
			};
			xhr.open("POST", url, true);
			xhr.setRequestHeader("Content-type", "application/sdp");
			xhr.send(offer.sdp);
		});
		await self.pc.setRemoteDescription(new RTCSessionDescription({ type: "answer", sdp: answer }));

		return self.__internal.parseId(url, offer.sdp, answer);
	};

	// Close the publisher.
	self.close = function () {
		self.pc && self.pc.close();
		self.pc = null;
	};

	// The callback when got local stream.
	// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
	self.ontrack = function (event) {
		// Add track to stream of SDK.
		self.stream.addTrack(event.track);
	};

	self.pc = new RTCPeerConnection(null);

	// To keep api consistent between player and publisher.
	// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
	// @see https://webrtc.org/getting-started/media-devices
	self.stream = new MediaStream();

	// Internal APIs.
	self.__internal = {
		parseId: (url, offer, answer) => {
			let sessionid = offer.substr(offer.indexOf("a=ice-ufrag:") + "a=ice-ufrag:".length);
			sessionid = sessionid.substr(0, sessionid.indexOf("\n") - 1) + ":";
			sessionid += answer.substr(answer.indexOf("a=ice-ufrag:") + "a=ice-ufrag:".length);
			sessionid = sessionid.substr(0, sessionid.indexOf("\n"));

			const a = document.createElement("a");
			a.href = url;
			return {
				sessionid: sessionid, // Should be ice-ufrag of answer:offer.
				simulator: a.protocol + "//" + a.host + "/rtc/v1/nack/"
			};
		}
	};

	// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
	self.pc.ontrack = function (event) {
		if (self.ontrack) {
			self.ontrack(event);
		}
	};

	return self;
}

// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
function SrsRtcFormatSenders(senders, kind) {
	var codecs = [];
	senders.forEach(function (sender) {
		var params = sender.getParameters();
		params &&
			params.codecs &&
			params.codecs.forEach(function (c) {
				if (kind && sender.track.kind !== kind) {
					return;
				}

				if (c.mimeType.indexOf("/red") > 0 || c.mimeType.indexOf("/rtx") > 0 || c.mimeType.indexOf("/fec") > 0) {
					return;
				}

				var s = "";

				s += c.mimeType.replace("audio/", "").replace("video/", "");
				s += ", " + c.clockRate + "HZ";
				if (sender.track.kind === "audio") {
					s += ", channels: " + c.channels;
				}
				s += ", pt: " + c.payloadType;

				codecs.push(s);
			});
	});
	return codecs.join(", ");
}

export default {
	SrsRtcPublisherAsync,
	SrsRtcPlayerAsync,
	SrsRtcWhipWhepAsync,
	SrsRtcFormatSenders,
	SrsError
};