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?
【解决方案】
三方库WebRTC适配HarmonyOS步骤如下:
-
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 -
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
- 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();
}
}
- 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();
}
}
- 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();
}
}
- 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();
}
}
- 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:
- 使用HarmonyOS NDK:通过Native API调用WebRTC C++库,利用FFI机制与ArkTS层交互
- 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)
- 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
- 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。

