/* eslint-disable */ import TRTC from 'trtc-js-sdk' class RtcClient { constructor(options) { this.sdkAppId_ = options.sdkAppId; this.userId_ = options.userId; this.userSig_ = options.userSig; this.roomId_ = options.roomId; this.isJoined_ = false; this.isPublished_ = false; this.localStream_ = null; this.remoteStreams_ = []; this.ready = false // check if browser is compatible with TRTC TRTC.checkSystemRequirements().then(result => { if (!result) { alert('Your browser is not compatible with TRTC! Please download Chrome M72+'); } }); } async join() { if (this.isJoined_) { console.warn('duplicate RtcClient.join() observed'); return; } // create a client for RtcClient this.client_ = TRTC.createClient({ mode: 'videoCall', // 实时通话模式 sdkAppId: this.sdkAppId_, userId: this.userId_, userSig: this.userSig_ }); // 处理 client 事件 this.handleEvents(); try { // join the room await this.client_.join({ roomId: this.roomId_ }); console.log('join room success'); this.isJoined_ = true; } catch (error) { window.dispatchEvent(new Event('leave')); console.warn('RtcClient.join join room fail:', JSON.stringify(error)) alert('进房失败') return; } this.localStream_.on('player-state-changed', event => { console.log(`local stream ${event.type} player is ${event.state}`); if (event.type === 'video' && event.state === 'PLAYING') { // dismiss the remote user UI placeholder } else if (event.type === 'video' && event.state === 'STOPPPED') { // show the remote user UI placeholder } }); try { this.localStream_.play('local') // 在id为 local 的 div 容器上播放本地音视频 } catch (e) { } // publish local stream by default after join the room await this.publish(); console.log('发布本地流成功!') } async leave() { if (!this.isJoined_) { console.warn('leave() - leave without join()d observed'); return; } if (this.isPublished_) { // ensure the local stream has been unpublished before leaving. await this.unpublish(true); } try { // leave the room await this.client_.leave(); this.isJoined_ = false; } catch (error) { location.reload(); } finally { // 停止本地流,关闭本地流内部的音视频播放器 this.localStream_.stop(); this.localStream_.close(); this.localStream_ = null; // 关闭本地流,释放摄像头和麦克风访问权限 } } async publish() { if (!this.isJoined_) { console.warn('publish() - please join() firstly'); return; } if (this.isPublished_) { console.warn('duplicate RtcClient.publish() observed'); return; } try { // 发布本地流 await this.client_.publish(this.localStream_); this.isPublished_ = true; } catch (error) { this.isPublished_ = false; } } async unpublish(isLeaving) { if (!this.isJoined_) { console.warn('unpublish() - please join() firstly'); return; } if (!this.isPublished_) { console.warn('RtcClient.unpublish() called but not published yet'); return; } try { // 停止发布本地流 await this.client_.unpublish(this.localStream_); this.isPublished_ = false; } catch (error) { if (!isLeaving) { console.warn('leaving the room because unpublish failure observed'); this.leave(); } } } muteLocalAudio() { this.localStream_.muteAudio(); } unmuteLocalAudio() { this.localStream_.unmuteAudio(); } muteLocalVideo() { this.localStream_.muteVideo(); } unmuteLocalVideo() { this.localStream_.unmuteVideo(); } async createLocalStream(options) { this.localStream_ = TRTC.createStream({ audio: options.audio, // 采集麦克风 video: options.video, // 采集摄像头 userId: this.userId_ // cameraId: getCameraId(), // microphoneId: getMicrophoneId() }); // 设置视频分辨率帧率和码率 this.localStream_.setVideoProfile('480p'); await this.localStream_.initialize(); } handleEvents() { // 处理 client 错误事件,错误均为不可恢复错误,建议提示用户后刷新页面 this.client_.on('error', () => { // alert(err); // location.reload(); }); // 处理用户被踢事件,通常是因为房间内有同名用户引起,这种问题一般是应用层逻辑错误引起的 // 应用层请尽量使用不同用户ID进房 this.client_.on('client-banned', () => { // location.reload(); }); // 远端用户进房通知 - 仅限主动推流用户 this.client_.on('peer-join', evt => { const userId = evt.userId; console.log('peer-join ' + userId); }); // 远端用户退房通知 - 仅限主动推流用户 this.client_.on('peer-leave', evt => { const userId = evt.userId; window.dispatchEvent(new Event('leave')); console.log('peer-leave ' + userId); }); // 处理远端流增加事件 this.client_.on('stream-added', evt => { const remoteStream = evt.stream; const id = remoteStream.getId(); const userId = remoteStream.getUserId(); console.log(`remote stream added: [${userId}] ID: ${id} type: ${remoteStream.getType()}`); console.log('subscribe to this remote stream'); // 远端流默认已订阅所有音视频,此处可指定只订阅音频或者音视频,不能仅订阅视频。 // 如果不想观看该路远端流,可调用 this.client_.unsubscribe(remoteStream) 取消订阅 this.client_.subscribe(remoteStream); }); // 远端流订阅成功事件 this.client_.on('stream-subscribed', evt => { const remoteStream = evt.stream; const id = remoteStream.getId(); this.remoteStreams_.push(remoteStream); try { document.getElementsByName('remote')[0].setAttribute('id', id); remoteStream.play(id); // 在指定的 div 容器上播放音视频 } catch(e) { } console.log('stream-subscribed ID: ', id); }); // 处理远端流被删除事件 this.client_.on('stream-removed', evt => { const remoteStream = evt.stream; const id = remoteStream.getId(); // 关闭远端流内部的音视频播放器 remoteStream.stop(); this.remoteStreams_ = this.remoteStreams_.filter(stream => { return stream.getId() !== id; }); console.log(`stream-removed ID: ${id} type: ${remoteStream.getType()}`); }); // 处理远端流更新事件,在音视频通话过程中,远端流音频或视频可能会有更新 this.client_.on('stream-updated', evt => { const remoteStream = evt.stream; console.log( 'type: ' + remoteStream.getType() + ' stream-updated hasAudio: ' + remoteStream.hasAudio() + ' hasVideo: ' + remoteStream.hasVideo() ); }); // 远端流音频或视频mute状态通知 this.client_.on('mute-audio', evt => { console.log(evt.userId + ' mute audio'); }); this.client_.on('unmute-audio', evt => { console.log(evt.userId + ' unmute audio'); }); this.client_.on('mute-video', evt => { console.log(evt.userId + ' mute video'); }); this.client_.on('unmute-video', evt => { console.log(evt.userId + ' unmute video'); }); // 信令通道连接状态通知 this.client_.on('connection-state-changed', evt => { console.log(`RtcClient state changed to ${evt.state} from ${evt.prevState}`); }); } } export default RtcClient