872 lines
32 KiB
JavaScript
872 lines
32 KiB
JavaScript
/**
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* 状态管理
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* 使用简单的 Observable 模式，可替换为 Redux/Vuex/Pinia
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*/
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import { mockCallSession, mockMessages } from './models.js';
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import { Signaling, WebSocketSignaling } from "../../module/signaling.js";// 信令管理
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import { RenderStreaming } from "../../module/renderstreaming.js"; // WebRTC连接管理
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import { getServerConfig, getRTCConfiguration } from "../js/config.js";//服务器配置和RTC配置
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import { showNotification } from './utils.js'; // 导入通知函数
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// 默认视频流尺寸
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const defaultStreamWidth = 1280;
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const defaultStreamHeight = 720;
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class CallStateManager {
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constructor() {
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const renderstreaming = null; // WebRTC连接管理实例
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const useWebSocket = null; // 是否使用WebSocket信令
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const connectionId = null; // 连接ID
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// 核心状态
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this.state = {
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session: {
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...mockCallSession,
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status: 'idle' // 初始状态为空闲
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},
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messages: [...mockMessages],
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isSidebarOpen: false,
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unreadCount: 0,
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localStream: null, // MediaStream 对象
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remoteStream: null // MediaStream 对象
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};
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// 监听器数组
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this.listeners = [];
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}
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// 订阅状态变化
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subscribe(callback) {
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this.listeners.push(callback);
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return () => {
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this.listeners = this.listeners.filter(cb => cb !== callback);
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};
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}
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// 通知所有监听器
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notify(changes) {
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this.listeners.forEach(cb => cb(this.state, changes));
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}
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// 初始化
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async init() {
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// 启动通话时长计时器
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this.durationInterval = setInterval(() => {
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this.state.session.duration++;
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this.notify({ type: 'DURATION_UPDATE', duration: this.state.session.duration });
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}, 1000);
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// 初始化配置
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await this.setupConfig();
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// 获取本地摄像头视频流
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await this.getLocalStream();
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}
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async setupConfig() {
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const res = await getServerConfig();
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this.useWebSocket = res.useWebSocket;
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}
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// 获取本地摄像头视频流
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async getLocalStream() {
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try {
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console.log('Requesting camera permission...');
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// 检查浏览器是否支持getUserMedia
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if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {
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console.error('getUserMedia is not supported');
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throw new Error('getUserMedia is not supported');
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}
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// 请求摄像头权限并获取媒体流
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const stream = await navigator.mediaDevices.getUserMedia({
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video: true,
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audio: true
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});
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console.log('Stream obtained successfully:', stream);
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console.log('Video tracks:', stream.getVideoTracks());
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console.log('Audio tracks:', stream.getAudioTracks());
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this.state.localStream = stream;
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this.state.session.localUser.mediaState.video = true;
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this.state.session.localUser.mediaState.audio = true;
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console.log('Local stream stored, notifying UI...');
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// 先通知视频流已获取
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this.notify({ type: 'LOCAL_STREAM_OBTAINED', stream });
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// 再通知媒体状态变化
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this.notify({ type: 'LOCAL_MEDIA_CHANGE', mediaType: 'video', value: true });
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this.notify({ type: 'LOCAL_MEDIA_CHANGE', mediaType: 'audio', value: true });
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// 发送媒体状态到服务器
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this.emitMediaStateChange();
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// 启动本地音频活动检测
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this.startLocalActivityDetection();
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} catch (error) {
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console.error('Error getting local stream:', error);
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// 如果获取视频失败，保持视频关闭状态
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this.state.session.localUser.mediaState.video = false;
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this.state.session.localUser.mediaState.audio = false;
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// 通知媒体状态变化
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this.notify({ type: 'LOCAL_MEDIA_CHANGE', mediaType: 'video', value: false });
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this.notify({ type: 'LOCAL_MEDIA_CHANGE', mediaType: 'audio', value: false });
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}
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}
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// 更新本地媒体状态
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async updateLocalMedia(mediaType, value) {
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// 如果是开启视频，重新获取摄像头资源
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if (mediaType === 'video' && value) {
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if (this.state.localStream) {
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// 停止当前的媒体流
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if (this.state.localStream) {
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this.state.localStream.getTracks().forEach(track => track.stop());
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}
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this.state.localStream = null;
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}
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// 请求摄像头权限并获取媒体流
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this.state.localStream = await navigator.mediaDevices.getUserMedia({
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video: true,
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audio: true
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});
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await this.getLocalStream();
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// 更新WebRTC连接中的视频轨道
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if (this.renderstreaming) {
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console.log('Updating video track in WebRTC connection');
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// 获取所有收发器
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const transceivers = this.renderstreaming.getTransceivers();
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console.log('All transceivers:', transceivers);
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// 查找现有的视频收发器
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const videoTransceivers = transceivers.filter(t => {
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return t.sender && t.sender.track && t.sender.track.kind === 'video';
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});
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console.log('Found video transceivers:', videoTransceivers);
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// 获取新的视频轨道
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const videoTracks = this.state.localStream.getVideoTracks();
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console.log('New video tracks:', videoTracks);
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if (videoTracks.length > 0) {
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const newVideoTrack = videoTracks[0];
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console.log('Using new video track:', newVideoTrack);
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if (videoTransceivers.length > 0) {
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// 替换现有的视频轨道
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for (const transceiver of videoTransceivers) {
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try {
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console.log('Replacing video track in transceiver:', transceiver);
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await transceiver.sender.replaceTrack(newVideoTrack);
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console.log('Successfully replaced video track');
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} catch (error) {
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console.error('Error replacing video track:', error);
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}
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}
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} else {
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// 添加新的视频收发器
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try {
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console.log('Adding new video transceiver');
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const transceiver = this.renderstreaming.addTransceiver(newVideoTrack, { direction: 'sendonly' });
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console.log('Added new video transceiver:', transceiver);
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} catch (error) {
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console.error('Error adding new video transceiver:', error);
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}
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}
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// 延迟设置编解码器偏好，确保收发器已完全创建
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setTimeout(() => {
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this.setCodecPreferences();
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}, 100);
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}
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}
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} else {
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// 直接更新媒体状态
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this.state.session.localUser.mediaState[mediaType] = value;
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this.notify({ type: 'LOCAL_MEDIA_CHANGE', mediaType, value });
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// 发送媒体状态到服务器
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this.emitMediaStateChange();
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}
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// 如果是关闭视频，释放摄像头资源
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if (mediaType === 'video' && !value && this.state.localStream) {
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this.state.localStream.getTracks().forEach(track => {
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if (track.kind === 'video') {
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track.stop();
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}
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});
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}
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// 如果是音频状态变化，控制本地音频轨道
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if (mediaType === 'audio' && this.state.localStream) {
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this.state.localStream.getTracks().forEach(track => {
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if (track.kind === 'audio') {
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track.enabled = value;
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}
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});
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}
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}
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/**
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* 设置WebRTC连接
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* @async
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* @returns {Promise<void>}
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*/
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async setUp(connectionId) {
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//TODO
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this.connectionId = connectionId; // 获取连接ID
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// 设置状态为连接中
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this.state.session.status = 'connecting';
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this.notify({ type: 'CALL_STATUS_CHANGE', status: 'connecting' });
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// 确保本地流已经初始化
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if (!this.state.localStream) {
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console.log('Local stream not available, waiting for initialization...');
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// 等待localStream初始化
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await new Promise((resolve) => {
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const checkStream = () => {
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if (this.state.localStream) {
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resolve();
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} else {
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setTimeout(checkStream, 100);
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}
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};
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checkStream();
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});
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}
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// 创建信令实例
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const signaling = this.useWebSocket ? new WebSocketSignaling() : new Signaling();
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const config = getRTCConfiguration(); // 获取RTC配置
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// 优化RTC配置，确保支持高分辨率
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config.peerConnectionOptions = {
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optional: [
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{ googCpuOveruseDetection: false }, // 禁用CPU过度使用检测
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{ googScreencastMinBitrate: 3000 } // 设置最小比特率
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]
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};
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this.renderstreaming = new RenderStreaming(signaling, config); // 创建WebRTC连接管理实例
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// 连接建立回调
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this.renderstreaming.onConnect = () => {
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// 连接建立后，更新状态为ongoing
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this.state.session.status = 'ongoing';
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this.notify({ type: 'CALL_STATUS_CHANGE', status: 'ongoing' });
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if (this.state.localStream) {
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const tracks = this.state.localStream.getTracks(); // 获取本地媒体轨道
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for (const track of tracks) {
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this.renderstreaming.addTransceiver(track, { direction: 'sendonly' }); // 添加发送轨道
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}
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this.setCodecPreferences(); // 设置编解码器偏好
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this.showStatsMessage(); // 显示统计信息
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} else {
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console.error('Local stream is not available');
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showNotification('本地视频流不可用', 'error');
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}
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};
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// 连接断开回调
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this.renderstreaming.onDisconnect = () => {
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this.hangUp(); // 挂断连接
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};
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// 轨道事件回调
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this.renderstreaming.onTrackEvent = (data) => {
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const direction = data.transceiver.direction;
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if (direction == "sendrecv" || direction == "recvonly") {
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if (this.state.remoteStream == null) {
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this.state.remoteStream = new MediaStream();
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}
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// 检查是否已经有相同类型的轨道
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const existingTracks = this.state.remoteStream.getTracks().filter(track => track.kind === data.track.kind);
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// 移除旧的轨道
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existingTracks.forEach(track => {
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this.state.remoteStream.removeTrack(track);
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console.log('Removed old track:', track.kind);
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});
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// 添加新的轨道
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this.state.remoteStream.addTrack(data.track);
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console.log('Added new track:', data.track.kind);
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// 通知UI远程流已更新
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this.notify({ type: 'REMOTE_STREAM_OBTAINED', stream: this.state.remoteStream });
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console.log('Notified UI about remote stream update');
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// 如果是音频轨道，启动远程音频活动检测
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if (data.track.kind === 'audio') {
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this.startRemoteActivityDetection();
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}
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} else if (direction == "sendonly") {
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// 本地发送轨道，启动本地音频活动检测
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if (data.track.kind === 'audio') {
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this.startLocalActivityDetection();
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}
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}
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};
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// 启动WebRTC连接
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await this.renderstreaming.start();
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await this.renderstreaming.createConnection(connectionId);
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// 启动网络质量检测
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this.startNetworkQualityDetection();
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// 启动本地音频活动检测
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this.startLocalActivityDetection();
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// 启动远端音频活动检测
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this.startRemoteActivityDetection();
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//模拟远端活动 (开发测试用)
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//this.simulateRemoteActivity();
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}
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/**
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* 挂断WebRTC连接
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* @async
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* @returns {Promise<void>}
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*/
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async hangUp() {
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this.clearStatsMessage(); // 清除统计信息
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this.stopNetworkQualityDetection(); // 停止网络质量检测
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console.log(`Disconnect peer on ${this.connectionId}.`);
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// 删除连接并停止WebRTC
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if (this.renderstreaming) {
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try {
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await this.renderstreaming.deleteConnection();
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await this.renderstreaming.stop();
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} catch (error) {
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console.error('Error during hangUp:', error);
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}
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this.renderstreaming = null;
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}
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this.connectionId = null;
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this.state.session.status = 'ended';
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this.notify({ type: 'CALL_ENDED' });
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}
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/**
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* 设置编解码器偏好
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*/
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setCodecPreferences() {
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/** @type {RTCRtpCodecCapability[] | null} */
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let selectedCodecs = null;
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// 获取视频编解码器能力
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const { codecs } = RTCRtpSender.getCapabilities('video');
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if (codecs && codecs.length > 0) {
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// 优先选择H.264编解码器
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const h264Codec = codecs.find(c => c.mimeType === 'video/H264');
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if (h264Codec) {
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selectedCodecs = [h264Codec];
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}
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}
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if (selectedCodecs == null) {
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return;
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}
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// 获取视频收发器并设置编解码器偏好
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if (this.renderstreaming) {
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const transceivers = this.renderstreaming.getTransceivers().filter(t => t.receiver.track.kind == "video");
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if (transceivers && transceivers.length > 0) {
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transceivers.forEach(t => t.setCodecPreferences(selectedCodecs));
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}
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}
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}
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// 更新远端媒体状态 (由 WebSocket 消息触发)
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updateRemoteMedia(mediaState) {
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this.state.session.remoteUser.mediaState = {
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...this.state.session.remoteUser.mediaState,
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...mediaState
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};
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this.notify({ type: 'REMOTE_MEDIA_CHANGE', mediaState });
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}
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// 添加消息
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addMessage(message) {
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this.state.messages.push(message);
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// 如果侧边栏关闭且不是自己发的，增加未读
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if (!this.state.isSidebarOpen && !message.isSelf) {
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this.state.unreadCount++;
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}
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this.notify({ type: 'NEW_MESSAGE', message, unreadCount: this.state.unreadCount });
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}
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// 切换侧边栏
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toggleSidebar() {
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this.state.isSidebarOpen = !this.state.isSidebarOpen;
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if (this.state.isSidebarOpen) {
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this.state.unreadCount = 0;
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}
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this.notify({ type: 'SIDEBAR_TOGGLE', isOpen: this.state.isSidebarOpen });
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}
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// 结束通话
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endCall() {
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clearInterval(this.durationInterval);
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this.state.session.status = 'ended';
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this.notify({ type: 'CALL_ENDED' });
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// 发送结束通话请求到服务器
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// [API_CALL: POST /api/call/:callId/leave]
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// [WEBSOCKET_EMIT: leave-call]
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}
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// 加入通话
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async joinCall(connectionId) {
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this.state.session.status = 'connecting';
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this.notify({ type: 'CALL_STATUS_CHANGE', status: 'connecting' });
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showNotification(`正在加入通话 (${connectionId})`);
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// 初始化
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await this.init();
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// 保存连接ID
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this.connectionId = connectionId;
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}
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// 创建通话
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async createCall() {
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this.state.session.status = 'connecting';
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this.notify({ type: 'CALL_STATUS_CHANGE', status: 'connecting' });
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showNotification('正在创建通话...');
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// 初始化
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await this.init();
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}
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// 模拟远端活动 (开发测试用)
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simulateRemoteActivity() {
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setInterval(() => {
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if (Math.random() > 0.7) {
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const isSpeaking = Math.random() > 0.5;
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this.updateRemoteMedia({ isSpeaking });
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}
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}, 800);
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}
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simulateNetworkChange() {
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// 模拟网络质量变化
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const qualities = ['good', 'fair', 'excellent', 'poor'];
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setInterval(() => {
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if (Math.random() > 0.8) {
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const quality = qualities[Math.floor(Math.random() * qualities.length)];
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this.state.session.remoteUser.networkQuality = quality;
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this.notify({ type: 'NETWORK_CHANGE', quality });
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}
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}, 5000);
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}
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// 真实网络质量检测
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async detectNetworkQuality() {
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if (!this.renderstreaming) {
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return;
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}
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try {
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const stats = await this.renderstreaming.getStats();
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if (!stats) {
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return;
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}
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let totalPacketsLost = 0;
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let totalPacketsReceived = 0;
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let inboundRTPCount = 0;
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let jitter = 0;
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let roundTripTime = 0;
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// 分析统计信息
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stats.forEach(report => {
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if (report.type === 'inbound-rtp' && report.mediaType === 'video') {
|
||
inboundRTPCount++;
|
||
|
||
// 计算丢包率
|
||
if (report.packetsLost !== undefined && report.packetsReceived !== undefined) {
|
||
totalPacketsLost += report.packetsLost;
|
||
totalPacketsReceived += report.packetsReceived;
|
||
}
|
||
|
||
// 获取抖动
|
||
if (report.jitter !== undefined) {
|
||
jitter = Math.max(jitter, report.jitter);
|
||
}
|
||
|
||
// 获取往返时间
|
||
if (report.roundTripTime !== undefined) {
|
||
roundTripTime = Math.max(roundTripTime, report.roundTripTime);
|
||
}
|
||
}
|
||
});
|
||
|
||
// 计算网络质量指标
|
||
let quality = 'excellent';
|
||
|
||
if (inboundRTPCount > 0) {
|
||
// 基于丢包率判断
|
||
const packetLossRate = totalPacketsReceived > 0 ? (totalPacketsLost / (totalPacketsLost + totalPacketsReceived)) : 0;
|
||
|
||
// 基于抖动判断
|
||
const jitterMs = jitter * 1000;
|
||
|
||
// 基于往返时间判断
|
||
const rttMs = roundTripTime * 1000;
|
||
|
||
// 综合评估网络质量
|
||
if (packetLossRate > 0.05 || jitterMs > 100 || rttMs > 300) {
|
||
quality = 'poor';
|
||
} else if (packetLossRate > 0.02 || jitterMs > 50 || rttMs > 150) {
|
||
quality = 'fair';
|
||
} else if (packetLossRate > 0.01 || jitterMs > 30 || rttMs > 100) {
|
||
quality = 'good';
|
||
} else {
|
||
quality = 'excellent';
|
||
}
|
||
}
|
||
|
||
// 更新网络质量状态
|
||
if (this.state.session.remoteUser.networkQuality !== quality) {
|
||
this.state.session.remoteUser.networkQuality = quality;
|
||
this.notify({ type: 'NETWORK_CHANGE', quality });
|
||
}
|
||
|
||
} catch (error) {
|
||
console.error('Error detecting network quality:', error);
|
||
}
|
||
}
|
||
// 真实音频活动检测 - 远端
|
||
startRemoteActivityDetection() {
|
||
// 检查是否有远端音频流
|
||
if (!this.state.remoteStream) {
|
||
return;
|
||
}
|
||
|
||
// 获取音频轨道
|
||
const audioTracks = this.state.remoteStream.getAudioTracks();
|
||
if (audioTracks.length === 0) {
|
||
return;
|
||
}
|
||
|
||
try {
|
||
// 创建音频上下文
|
||
const audioContext = new (window.AudioContext || window.webkitAudioContext)();
|
||
|
||
// 创建媒体流源
|
||
const source = audioContext.createMediaStreamSource(this.state.remoteStream);
|
||
|
||
// 创建音频分析器
|
||
const analyser = audioContext.createAnalyser();
|
||
analyser.fftSize = 256;
|
||
|
||
// 连接音频节点
|
||
source.connect(analyser);
|
||
|
||
// 创建数据缓冲区
|
||
const dataArray = new Uint8Array(analyser.frequencyBinCount);
|
||
|
||
// 检测参数
|
||
const threshold = 15; // 音频电平阈值
|
||
const debounceTime = 500; // 防抖时间
|
||
let isSpeaking = false;
|
||
let lastActivityTime = 0;
|
||
|
||
// 音频活动检测循环
|
||
const detectActivity = () => {
|
||
if (!this.state.remoteStream || !this.renderstreaming) {
|
||
return;
|
||
}
|
||
|
||
// 获取时域数据
|
||
analyser.getByteTimeDomainData(dataArray);
|
||
|
||
// 计算音频电平
|
||
let sum = 0;
|
||
for (let i = 0; i < dataArray.length; i++) {
|
||
// 转换为振幅 (0-255 → -128-127)
|
||
const amplitude = dataArray[i] - 128;
|
||
sum += amplitude * amplitude;
|
||
}
|
||
const rms = Math.sqrt(sum / dataArray.length);
|
||
const level = rms / 128; // 归一化到 0-1
|
||
|
||
// 检测说话状态
|
||
const currentTime = Date.now();
|
||
if (level > threshold / 100) {
|
||
// 检测到说话
|
||
lastActivityTime = currentTime;
|
||
if (!isSpeaking) {
|
||
isSpeaking = true;
|
||
this.updateRemoteMedia({ isSpeaking: true });
|
||
}
|
||
} else if (isSpeaking && currentTime - lastActivityTime > debounceTime) {
|
||
// 停止说话
|
||
isSpeaking = false;
|
||
this.updateRemoteMedia({ isSpeaking: false });
|
||
}
|
||
|
||
// 继续检测
|
||
if (this.state.session.status === 'ongoing') {
|
||
requestAnimationFrame(detectActivity);
|
||
}
|
||
};
|
||
|
||
// 开始检测
|
||
detectActivity();
|
||
|
||
console.log('Remote activity detection started');
|
||
|
||
} catch (error) {
|
||
console.error('Error starting remote activity detection:', error);
|
||
}
|
||
}
|
||
|
||
// 真实音频活动检测 - 本地
|
||
startLocalActivityDetection() {
|
||
// 检查是否有本地音频流
|
||
if (!this.state.localStream) {
|
||
return;
|
||
}
|
||
|
||
// 获取音频轨道
|
||
const audioTracks = this.state.localStream.getAudioTracks();
|
||
if (audioTracks.length === 0) {
|
||
return;
|
||
}
|
||
|
||
try {
|
||
// 创建音频上下文
|
||
const audioContext = new (window.AudioContext || window.webkitAudioContext)();
|
||
|
||
// 创建媒体流源
|
||
const source = audioContext.createMediaStreamSource(this.state.localStream);
|
||
|
||
// 创建音频分析器
|
||
const analyser = audioContext.createAnalyser();
|
||
analyser.fftSize = 256;
|
||
|
||
// 连接音频节点
|
||
source.connect(analyser);
|
||
|
||
// 创建数据缓冲区
|
||
const dataArray = new Uint8Array(analyser.frequencyBinCount);
|
||
|
||
// 检测参数
|
||
const threshold = 15; // 音频电平阈值
|
||
const debounceTime = 500; // 防抖时间
|
||
let isSpeaking = false;
|
||
let lastActivityTime = 0;
|
||
|
||
// 音频活动检测循环
|
||
const detectActivity = () => {
|
||
if (!this.state.localStream || !this.renderstreaming) {
|
||
return;
|
||
}
|
||
|
||
// 获取时域数据
|
||
analyser.getByteTimeDomainData(dataArray);
|
||
|
||
// 计算音频电平
|
||
let sum = 0;
|
||
for (let i = 0; i < dataArray.length; i++) {
|
||
// 转换为振幅 (0-255 → -128-127)
|
||
const amplitude = dataArray[i] - 128;
|
||
sum += amplitude * amplitude;
|
||
}
|
||
const rms = Math.sqrt(sum / dataArray.length);
|
||
const level = rms / 128; // 归一化到 0-1
|
||
|
||
// 检测说话状态
|
||
const currentTime = Date.now();
|
||
if (level > threshold / 100) {
|
||
// 检测到说话
|
||
lastActivityTime = currentTime;
|
||
if (!isSpeaking) {
|
||
isSpeaking = true;
|
||
this.state.session.localUser.mediaState.isSpeaking = true;
|
||
this.notify({ type: 'LOCAL_MEDIA_CHANGE', mediaType: 'isSpeaking', value: true });
|
||
// 发送媒体状态到服务器
|
||
this.emitMediaStateChange();
|
||
}
|
||
} else if (isSpeaking && currentTime - lastActivityTime > debounceTime) {
|
||
// 停止说话
|
||
isSpeaking = false;
|
||
this.state.session.localUser.mediaState.isSpeaking = false;
|
||
this.notify({ type: 'LOCAL_MEDIA_CHANGE', mediaType: 'isSpeaking', value: false });
|
||
// 发送媒体状态到服务器
|
||
this.emitMediaStateChange();
|
||
}
|
||
|
||
// 继续检测
|
||
if (this.state.session.status === 'ongoing') {
|
||
requestAnimationFrame(detectActivity);
|
||
}
|
||
};
|
||
|
||
// 开始检测
|
||
detectActivity();
|
||
|
||
console.log('Local activity detection started');
|
||
|
||
} catch (error) {
|
||
console.error('Error starting local activity detection:', error);
|
||
}
|
||
}
|
||
// 启动网络质量检测
|
||
startNetworkQualityDetection() {
|
||
// 每3秒检测一次网络质量
|
||
this.networkQualityInterval = setInterval(() => {
|
||
this.detectNetworkQuality();
|
||
//this.simulateNetworkChange();
|
||
}, 3000);
|
||
}
|
||
|
||
// 停止网络质量检测
|
||
stopNetworkQualityDetection() {
|
||
if (this.networkQualityInterval) {
|
||
clearInterval(this.networkQualityInterval);
|
||
this.networkQualityInterval = null;
|
||
}
|
||
}
|
||
|
||
// 发送媒体状态到服务器
|
||
emitMediaStateChange() {
|
||
const payload = {
|
||
userId: this.state.session.localUser.id,
|
||
...this.state.session.localUser.mediaState
|
||
};
|
||
console.log('[WebSocket Emit] media-state-changed:', payload);
|
||
// socket.emit('media-state-changed', payload);
|
||
}
|
||
|
||
// 显示统计信息
|
||
async showStatsMessage() {
|
||
console.log('Showing stats message');
|
||
|
||
// 立即执行一次网络质量检测
|
||
await this.detectNetworkQuality();
|
||
|
||
// 定期显示详细统计信息
|
||
this.statsInterval = setInterval(async () => {
|
||
if (!this.renderstreaming) {
|
||
return;
|
||
}
|
||
|
||
try {
|
||
const stats = await this.renderstreaming.getStats();
|
||
if (!stats) {
|
||
return;
|
||
}
|
||
|
||
let statsSummary = {
|
||
video: {
|
||
packetsLost: 0,
|
||
packetsReceived: 0,
|
||
bytesReceived: 0,
|
||
jitter: 0,
|
||
roundTripTime: 0,
|
||
fps: 0,
|
||
bitrate: 0
|
||
},
|
||
audio: {
|
||
packetsLost: 0,
|
||
packetsReceived: 0,
|
||
bytesReceived: 0,
|
||
jitter: 0
|
||
}
|
||
};
|
||
|
||
// 分析统计信息
|
||
stats.forEach(report => {
|
||
if (report.type === 'inbound-rtp') {
|
||
if (report.mediaType === 'video') {
|
||
statsSummary.video.packetsLost = report.packetsLost || 0;
|
||
statsSummary.video.packetsReceived = report.packetsReceived || 0;
|
||
statsSummary.video.bytesReceived = report.bytesReceived || 0;
|
||
statsSummary.video.jitter = report.jitter || 0;
|
||
statsSummary.video.roundTripTime = report.roundTripTime || 0;
|
||
statsSummary.video.fps = report.framesPerSecond || 0;
|
||
|
||
// 计算视频比特率 (kbps)
|
||
if (report.bytesReceived && report.timestamp) {
|
||
const duration = report.timestamp / 1000; // 转换为秒
|
||
statsSummary.video.bitrate = duration > 0 ? Math.round((report.bytesReceived * 8) / (duration * 1000)) : 0;
|
||
}
|
||
} else if (report.mediaType === 'audio') {
|
||
statsSummary.audio.packetsLost = report.packetsLost || 0;
|
||
statsSummary.audio.packetsReceived = report.packetsReceived || 0;
|
||
statsSummary.audio.bytesReceived = report.bytesReceived || 0;
|
||
statsSummary.audio.jitter = report.jitter || 0;
|
||
}
|
||
}
|
||
});
|
||
|
||
// 输出详细统计信息
|
||
console.log('=== WebRTC Statistics ===');
|
||
console.log(`Network Quality: ${this.state.session.remoteUser.networkQuality}`);
|
||
console.log('Video Stats:', {
|
||
'Packets Lost': statsSummary.video.packetsLost,
|
||
'Packets Received': statsSummary.video.packetsReceived,
|
||
'Packet Loss Rate': statsSummary.video.packetsReceived > 0 ?
|
||
`${((statsSummary.video.packetsLost / (statsSummary.video.packetsLost + statsSummary.video.packetsReceived)) * 100).toFixed(2)}%` : '0%',
|
||
'Jitter': `${(statsSummary.video.jitter * 1000).toFixed(2)}ms`,
|
||
'Round Trip Time': `${(statsSummary.video.roundTripTime * 1000).toFixed(2)}ms`,
|
||
'FPS': statsSummary.video.fps.toFixed(1),
|
||
'Bitrate': `${statsSummary.video.bitrate}kbps`
|
||
});
|
||
console.log('Audio Stats:', {
|
||
'Packets Lost': statsSummary.audio.packetsLost,
|
||
'Packets Received': statsSummary.audio.packetsReceived,
|
||
'Packet Loss Rate': statsSummary.audio.packetsReceived > 0 ?
|
||
`${((statsSummary.audio.packetsLost / (statsSummary.audio.packetsLost + statsSummary.audio.packetsReceived)) * 100).toFixed(2)}%` : '0%',
|
||
'Jitter': `${(statsSummary.audio.jitter * 1000).toFixed(2)}ms`
|
||
});
|
||
console.log('========================');
|
||
|
||
} catch (error) {
|
||
console.error('Error showing stats message:', error);
|
||
}
|
||
}, 5000); // 每5秒更新一次统计信息
|
||
}
|
||
|
||
// 清除统计信息
|
||
clearStatsMessage() {
|
||
console.log('Clearing stats message');
|
||
|
||
// 清理统计信息定时器
|
||
if (this.statsInterval) {
|
||
clearInterval(this.statsInterval);
|
||
this.statsInterval = null;
|
||
}
|
||
}
|
||
|
||
// Getters
|
||
getState() { return this.state; }
|
||
getLocalUser() { return this.state.session.localUser; }
|
||
getRemoteUser() { return this.state.session.remoteUser; }
|
||
getMessages() { return this.state.messages; }
|
||
getConnectionId() { return this.connectionId; }
|
||
}
|
||
|
||
// 创建单例实例
|
||
const store = new CallStateManager();
|
||
|
||
// 页面卸载前清理
|
||
window.addEventListener('beforeunload', async () => {
|
||
if (!store.renderstreaming)
|
||
return;
|
||
await store.renderstreaming.stop(); // 停止WebRTC连接
|
||
}, true);
|
||
export default store;
|