HarmonyOS鸿蒙Next中如何实现WebRTC版本适配?

HarmonyOS鸿蒙Next中如何实现WebRTC版本适配?

5 回复

【解决方案】

三方库WebRTC适配HarmonyOS步骤如下:

  1. HarmonyOS版本WebRTC编译。对WebRTC源码进行交叉编译,参考WebRTC编译指导。arm64-v8a架构编译命令和参数示意如下:

    gn gen out/arm64 --args='is_clang=true target_os="ohos" target_cpu="arm64" ohos_extra_ldflags="-static-libstdc++" is_official_build=true ohos_sdk_native_root="path/to/ohos/sdk"'
    ninja -C out/arm64
    
  2. 工程引用HarmonyOS WebRTC。编译生成相应的HarmonyOS动态库libohos_webrtc.so,arm64-v8a架构将so库放在ohos_webrtc/libs/arm64-v8a目录下。

HarmonyOS WebRTC适配和使用过程中遇到问题,可参考WebRTC适配/使用时遇到问题总结

【背景知识】

WebRTC(Web Real-Time Communications)是一项实时通讯技术,它允许网络应用或者站点,在不借助中间媒介的情况下,建立浏览器之间点对点(Peer-to-Peer)的连接,实现视频流和(或)音频流等任意数据的传输。

更多关于HarmonyOS鸿蒙Next中如何实现WebRTC版本适配?的实战系列教程也可以访问 https://www.itying.com/category-93-b0.html


  1. 音频采集与播放
// services/AudioService.ts

import { audio } from '@kit.AudioKit';
import { BusinessError } from '@kit.BasicServicesKit';

export class AudioService {
  private audioCapturer?: audio.AudioCapturer;
  private audioRenderer?: audio.AudioRenderer;

  // 初始化音频采集(麦克风)
  async initAudioCapture(): Promise<void> {
    let audioStreamInfo: audio.AudioStreamInfo = {
      samplingRate: audio.AudioSamplingRate.SAMPLE_RATE_48000,
      channels: audio.AudioChannel.CHANNEL_2,
      sampleFormat: audio.AudioSampleFormat.SAMPLE_FORMAT_S16LE,
      encodingType: audio.AudioEncodingType.ENCODING_TYPE_RAW
    };

    let audioCapturerInfo: audio.AudioCapturerInfo = {
      source: audio.SourceType.SOURCE_TYPE_VOICE_COMMUNICATION, // VoIP 通话
      capturerFlags: 0
    };

    let audioCapturerOptions: audio.AudioCapturerOptions = {
      streamInfo: audioStreamInfo,
      capturerInfo: audioCapturerInfo
    };

    this.audioCapturer = await audio.createAudioCapturer(audioCapturerOptions);

    // 监听音频数据
    this.audioCapturer.on('readData', (buffer: ArrayBuffer) => {
      // 编码并发送给对端
      this.sendAudioData(buffer);
    });
  }

  // 初始化音频播放(扬声器)
  async initAudioRenderer(): Promise<void> {
    let audioStreamInfo: audio.AudioStreamInfo = {
      samplingRate: audio.AudioSamplingRate.SAMPLE_RATE_48000,
      channels: audio.AudioChannel.CHANNEL_2,
      sampleFormat: audio.AudioSampleFormat.SAMPLE_FORMAT_S16LE,
      encodingType: audio.AudioEncodingType.ENCODING_TYPE_RAW
    };

    let audioRendererInfo: audio.AudioRendererInfo = {
      usage: audio.StreamUsage.STREAM_USAGE_VOICE_COMMUNICATION,
      rendererFlags: 0
    };

    let audioRendererOptions: audio.AudioRendererOptions = {
      streamInfo: audioStreamInfo,
      rendererInfo: audioRendererInfo
    };

    this.audioRenderer = await audio.createAudioRenderer(audioRendererOptions);

    // 监听音频写入请求
    this.audioRenderer.on('writeData', (buffer: ArrayBuffer) => {
      // 从网络接收的数据写入到播放缓冲区
      return audio.AudioDataCallbackResult.VALID;
    });
  }

  // 开始采集
  async startCapture(): Promise<void> {
    await this.audioCapturer?.start();
  }

  // 开始播放
  async startRenderer(): Promise<void> {
    await this.audioRenderer?.start();
  }

  // 发送音频数据到对端
  private sendAudioData(buffer: ArrayBuffer): void {
    // TODO: 编码后通过网络发送
  }

  // 释放资源
  async release(): Promise<void> {
    await this.audioCapturer?.stop();
    await this.audioCapturer?.release();
    await this.audioRenderer?.stop();
    await this.audioRenderer?.release();
  }
}
  1. 视频采集与预览
// services/VideoService.ts

import { camera } from '@kit.CameraKit';
import { BusinessError } from '@kit.BasicServicesKit';

export class VideoService {
  private cameraManager?: camera.CameraManager;
  private cameraInput?: camera.CameraInput;
  private previewOutput?: camera.PreviewOutput;
  private videoOutput?: camera.VideoOutput;
  private session?: camera.VideoSession;

  // 初始化摄像头
  async initCamera(surfaceId: string, context: Context): Promise<void> {
    // 获取相机管理器
    this.cameraManager = camera.getCameraManager(context);
    // 获取摄像头列表
    let cameras = this.cameraManager.getSupportedCameras();
    if (cameras.length === 0) {
      throw new Error('未找到摄像头');
    }

    // 创建相机输入
    this.cameraInput = this.cameraManager.createCameraInput(cameras[0]);
    await this.cameraInput.open();

    // 获取支持的输出能力
    let capability = this.cameraManager.getSupportedOutputCapability(
      cameras[0],
      camera.SceneMode.NORMAL_VIDEO
    );

    // 创建预览输出
    let previewProfile = capability.previewProfiles[0];
    this.previewOutput = this.cameraManager.createPreviewOutput(
      previewProfile,
      surfaceId
    );

    // 创建录像输出(用于获取视频数据)
    let videoProfile = capability.videoProfiles[0];
    this.videoOutput = this.cameraManager.createVideoOutput(videoProfile);

    // 创建会话
    let sessionObj = this.cameraManager.createSession(camera.SceneMode.NORMAL_VIDEO);
    this.session = sessionObj as camera.VideoSession;

    // 配置会话
    this.session.beginConfig();
    this.session.addInput(this.cameraInput);
    this.session.addOutput(this.previewOutput);
    this.session.addOutput(this.videoOutput);
    await this.session.commitConfig();
  }

  // 开始预览
  async startPreview(): Promise<void> {
    await this.session?.start();
  }

  // 停止预览
  async stopPreview(): Promise<void> {
    await this.session?.stop();
  }

  // 释放资源
  async release(): Promise<void> {
    await this.session?.release();
    await this.cameraInput?.close();
    await this.previewOutput?.release();
    await this.videoOutput?.release();
  }
}
  1. 信令服务(WebSocket)
// services/SignalingService.ts

import { webSocket } from '@kit.NetworkKit';
import { BusinessError } from '@kit.BasicServicesKit';

export class SignalingService {
  private ws?: webSocket.WebSocket;
  private signalingServer: string = 'wss://your-signaling-server.com';

  // 连接信令服务器
  connect(): void {
    this.ws = webSocket.createWebSocket();

    // 监听连接成功
    this.ws.on('open', () => {
      console.info('信令服务器连接成功');
    });

    // 监听消息
    this.ws.on('message', (err: BusinessError, value: string | ArrayBuffer) => {
      if (!err) {
        let message = JSON.parse(value as string);
        this.handleSignalingMessage(message);
      }
    });

    // 监听错误
    this.ws.on('error', (err: BusinessError) => {
      console.error('WebSocket 错误:', err);
    });

    // 连接
    this.ws.connect(this.signalingServer);
  }

  // 发送信令消息
  send(message: object): void {
    this.ws?.send(JSON.stringify(message));
  }

  // 处理信令消息
  private handleSignalingMessage(message: any): void {
    switch (message.type) {
      case 'offer':
        // 处理 SDP Offer
        this.handleOffer(message.sdp);
        break;
      case 'answer':
        // 处理 SDP Answer
        this.handleAnswer(message.sdp);
        break;
      case 'candidate':
        // 处理 ICE Candidate
        this.handleCandidate(message.candidate);
        break;
    }
  }

  private handleOffer(sdp: string): void {
    // TODO: 处理 SDP Offer
  }

  private handleAnswer(sdp: string): void {
    // TODO: 处理 SDP Answer
  }

  private handleCandidate(candidate: any): void {
    // TODO: 处理 ICE Candidate
  }

  // 断开连接
  disconnect(): void {
    this.ws?.close();
  }
}
  1. WebRTC 管理类
// services/WebRTCManager.ts

import { AudioService } from './AudioService';
import { VideoService } from './VideoService';
import { SignalingService } from './SignalingService';

export class WebRTCManager {
  private audioService: AudioService;
  private videoService: VideoService;
  private signalingService: SignalingService;

  constructor() {
    this.audioService = new AudioService();
    this.videoService = new VideoService();
    this.signalingService = new SignalingService();
  }

  // 初始化 WebRTC
  async init(surfaceId: string, context: Context): Promise<void> {
    // 初始化音频
    await this.audioService.initAudioCapture();
    await this.audioService.initAudioRenderer();
    // 初始化视频
    await this.videoService.initCamera(surfaceId, context);
    // 连接信令服务器
    this.signalingService.connect();
  }

  // 开始通话
  async startCall(): Promise<void> {
    await this.audioService.startCapture();
    await this.audioService.startRenderer();
    await this.videoService.startPreview();
    // 发送 Offer
    // TODO: 实现 SDP 协商
  }

  // 结束通话
  async endCall(): Promise<void> {
    await this.audioService.release();
    await this.videoService.release();
    this.signalingService.disconnect();
  }
}
  1. 使用示例
// pages/WebRTCCall.ets

import { WebRTCManager } from '../services/WebRTCManager';

@Entry
@Component
struct WebRTCCallPage {
  private xComponentCtl: XComponentController = new XComponentController();
  private surfaceId: string = '';
  private webrtcManager: WebRTCManager = new WebRTCManager();
  private context: Context = this.getUIContext().getHostContext();

  build() {
    Column() {
      // 视频预览组件
      XComponent({
        id: 'videoPreview',
        type: XComponentType.SURFACE,
        controller: this.xComponentCtl
      })
        .onLoad(() => {
          this.surfaceId = this.xComponentCtl.getXComponentSurfaceId();
          this.webrtcManager.init(this.surfaceId, this.context);
        })
        .width('100%')
        .height('60%')

      // 控制按钮
      Row() {
        Button('开始通话')
          .onClick(() => {
            this.webrtcManager.startCall();
          })
        Button('结束通话')
          .onClick(() => {
            this.webrtcManager.endCall();
          })
      }
      .padding(20)
    }
  }
}

web-rtc # Web 中的 WebRTC audio-call-development # 音频通话开发 camera-preview # 摄像头预览 websocket-connection # WebSocket 信令 avcodec # 音视频编解码 socket-connection.md # Socket 数据传输

直接在 HarmonyOS 上实现 WebRTC 可能需要借助于一些特定的库和框架来完成

在HarmonyOS Next中实现WebRTC版本适配,需使用ArkTS语言调用鸿蒙原生WebRTC API。通过@ohos.multimedia.media库中的AVRecorder和AVPlayer组件处理音视频流,利用DeviceManager进行设备管理。适配时需关注鸿蒙系统版本与WebRTC规范的兼容性,调用createAVRecorder()方法创建实例,通过supportedFormats属性检查编解码支持。使用NetworkCapabilities API处理网络传输,确保SDP协商符合鸿蒙实现规范。

在HarmonyOS Next中实现WebRTC适配,建议通过以下方式:

  1. 使用HarmonyOS NDK:通过Native API调用WebRTC C++库,利用FFI机制与ArkTS层交互
  2. 模块化改造
    • 将WebRTC媒体引擎封装为Native SDK
    • 通过NAPI暴露关键接口(createPeerConnection、createVideoTrack等)
  3. 硬件适配
    • 使用HarmonyOS相机接口替换原视频采集模块
    • 适配AudioManager进行音频路由管理
  4. 网络层适配
    • 基于HarmonyOS网络框架实现ICE候选收集
    • 使用系统Socket服务替代原有网络模块

注意需要重新编译WebRTC源码,针对HarmonyOS架构进行交叉编译,并解决依赖库的兼容性问题。建议参考OpenHarmony社区已有的多媒体框架实现方案。

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