Flutter WebRTC通信插件medea_flutter_webrtc的使用

发布于 1周前 作者 wuwangju 来自 Flutter

Flutter WebRTC通信插件 medea_flutter_webrtc 的使用

简介

medea_flutter_webrtc 是一个用于 FlutterWebRTC 插件,设计并应用于 Medea Jason WebRTC 客户端,构建于预编译的 libwebrtc 二进制文件之上。

支持的平台

  • macOS 10.11+
  • Linux (需要 PulseAudio 和 X11 进行屏幕共享)
  • Windows 7+
  • Android 24+
  • iOS 13+
  • Web(部分支持)

使用步骤

添加依赖

pubspec.yaml 文件中添加 medea_flutter_webrtc 作为依赖:

dependencies:
  medea_flutter_webrtc: ^最新版本号

请确保将 ^最新版本号 替换为实际的最新版本号。可以从 pub.dev 获取最新的版本信息。

Android 配置

<project_root>/android/app/src/main/AndroidManifest.xml 文件中添加以下权限:

<uses-feature android:name="android.hardware.camera" />
<uses-feature android:name="android.hardware.camera.autofocus" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
<uses-permission android:name="android.permission.CAMERA" />
<uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" />
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />

如果需要使用蓝牙设备(如耳机),还需添加以下权限:

<uses-permission android:name="android.permission.BLUETOOTH" android:maxSdkVersion="30" />
<uses-permission android:name="android.permission.BLUETOOTH_ADMIN" android:maxSdkVersion="30" />
<uses-permission android:name="android.permission.BLUETOOTH_CONNECT" />

同时,在 build.gradle 中设置 Java 8 兼容性:

android {
    //...
    compileOptions {
        sourceCompatibility JavaVersion.VERSION_1_8
        targetCompatibility JavaVersion.VERSION_1_8
    }
}

iOS 配置

<project_root>/ios/Runner/Info.plist 文件中添加以下条目:

<key>NSCameraUsageDescription</key>
<string>$(PRODUCT_NAME) Camera Usage!</string>
<key>NSMicrophoneUsageDescription</key>
<string>$(PRODUCT_NAME) Microphone Usage!</string>

这些条目允许应用访问相机和麦克风。

示例代码

下面是一个简单的示例应用,展示了如何使用 medea_flutter_webrtc 插件的基本功能。

import 'package:flutter/material.dart';
import 'package:medea_flutter_webrtc/medea_flutter_webrtc.dart';

void main() async {
  await initFfiBridge();
  runApp(const MyApp());
}

class MyApp extends StatefulWidget {
  const MyApp({super.key});

  [@override](/user/override)
  State<MyApp> createState() => _MyAppState();
}

class _MyAppState extends State<MyApp> {
  late List<RouteItem> items;

  [@override](/user/override)
  void initState() {
    super.initState();
    _initItems();
  }

  ListBody _buildRow(context, item) {
    return ListBody(
      children: <Widget>[
        ListTile(
          title: Text(item.title),
          onTap: () => item.push(context),
          trailing: const Icon(Icons.arrow_right),
        ),
        const Divider()
      ],
    );
  }

  [@override](/user/override)
  Widget build(BuildContext context) {
    return MaterialApp(
      home: Scaffold(
        appBar: AppBar(title: const Text('Flutter-WebRTC example')),
        body: ListView.builder(
          shrinkWrap: true,
          padding: const EdgeInsets.all(0.0),
          itemCount: items.length,
          itemBuilder: (context, i) {
            return _buildRow(context, items[i]);
          },
        ),
      ),
      debugShowCheckedModeBanner: false,
    );
  }

  void _initItems() {
    items = <RouteItem>[
      RouteItem(
        title: 'GetUserMedia',
        push: (BuildContext context) {
          Navigator.push(
            context,
            MaterialPageRoute(
              builder: (BuildContext context) => const GetUserMediaSample(),
            ),
          );
        },
      ),
      RouteItem(
        title: 'GetDisplayMedia',
        push: (BuildContext context) {
          Navigator.push(
            context,
            MaterialPageRoute(
              builder: (BuildContext context) => const GetDisplayMediaSample(),
            ),
          );
        },
      ),
      RouteItem(
        title: 'LoopBack Sample',
        push: (BuildContext context) {
          Navigator.push(
            context,
            MaterialPageRoute(
              builder: (BuildContext context) => const Loopback(),
            ),
          );
        },
      ),
      RouteItem(
        title: 'getSources',
        push: (BuildContext context) {
          Navigator.push(
            context,
            MaterialPageRoute(
              builder: (BuildContext context) => const GetSourcesSample(),
            ),
          );
        },
      ),
      RouteItem(
        title: 'Basic RtcPeerConnection',
        push: (BuildContext context) {
          Navigator.push(
            context,
            MaterialPageRoute(
              builder: (BuildContext context) => const PeerConnectionSample(),
            ),
          );
        },
      ),
      RouteItem(
        title: 'onDeviceChange notifier',
        push: (BuildContext context) {
          Navigator.push(
            context,
            MaterialPageRoute(
              builder: (BuildContext context) =>
                  const OnDeviceChangeNotifierSample(),
            ),
          );
        },
      ),
      RouteItem(
        title: 'Video Codec Info',
        push: (BuildContext context) {
          Navigator.push(
            context,
            MaterialPageRoute(
              builder: (BuildContext context) => const VideoCodecInfoSample(),
            ),
          );
        },
      ),
    ];
  }
}

class RouteItem {
  final String title;
  final Function(BuildContext) push;

  RouteItem({required this.title, required this.push});
}

更多关于Flutter WebRTC通信插件medea_flutter_webrtc的使用的实战系列教程也可以访问 https://www.itying.com/category-92-b0.html

1 回复

更多关于Flutter WebRTC通信插件medea_flutter_webrtc的使用的实战系列教程也可以访问 https://www.itying.com/category-92-b0.html


当然,下面是一个关于如何在Flutter项目中使用media_flutter_webrtc插件进行WebRTC通信的示例代码。media_flutter_webrtc是一个用于Flutter的WebRTC库,它允许你实现视频和音频的实时通信。

首先,确保你已经在pubspec.yaml文件中添加了media_flutter_webrtc依赖:

dependencies:
  flutter:
    sdk: flutter
  media_flutter_webrtc: ^0.6.10  # 请检查最新版本号并更新

然后运行flutter pub get来安装依赖。

接下来,我们需要进行一些基本的设置来初始化WebRTC环境,并创建一个简单的视频通话应用。以下是一个基本的示例代码,包括信令服务器的设置(这里假设你有一个WebSocket信令服务器来处理SDP和ICE候选信息的交换)。

main.dart

import 'package:flutter/material.dart';
import 'package:media_flutter_webrtc/media_flutter_webrtc.dart';
import 'dart:async';
import 'dart:typed_data';
import 'package:web_socket_channel/web_socket_channel.dart';

void main() {
  runApp(MyApp());
}

class MyApp extends StatelessWidget {
  @override
  Widget build(BuildContext context) {
    return MaterialApp(
      home: WebRTCDemo(),
    );
  }
}

class WebRTCDemo extends StatefulWidget {
  @override
  _WebRTCDemoState createState() => _WebRTCDemoState();
}

class _WebRTCDemoState extends State<WebRTCDemo> {
  RTCPeerConnection? peerConnection;
  VideoRenderer? localRenderer;
  VideoRenderer? remoteRenderer;
  WebSocketChannel? channel;

  @override
  void initState() {
    super.initState();
    initRenderers();
    initPeerConnection();
    connectToSignalingServer();
  }

  Future<void> initRenderers() async {
    localRenderer = createVideoRenderer();
    await localRenderer!.initialize();
    remoteRenderer = createVideoRenderer();
    await remoteRenderer!.initialize();
  }

  Future<void> initPeerConnection() async {
    Map<String, dynamic> configuration = {
      "iceServers": [
        {"urls": "stun:stun.l.google.com:19302"},
      ]
    };
    peerConnection = await createPeerConnection(configuration);

    peerConnection!.onTrack = (RTCTrackEvent event) {
      remoteRenderer!.srcObject = event.streams[0];
    };

    peerConnection!.onIceCandidate = (RTCIceCandidate candidate) {
      if (candidate != null) {
        sendMessage(jsonEncode({"candidate": candidate.toMap()}));
      }
    };
  }

  Future<void> connectToSignalingServer() async {
    channel = WebSocketChannel.connect(Uri.parse('wss://your-signaling-server-url'));
    channel!.stream.listen((message) {
      Map<String, dynamic> data = jsonDecode(message);
      String type = data['type']!;

      if (type == 'offer') {
        peerConnection!.setRemoteDescription(
            RTCSessionDescription(data['sdp']!, data['type']!)).then((_) {
          return peerConnection!.createAnswer();
        }).then((RTCSessionDescription answer) {
          return peerConnection!.setLocalDescription(answer);
        }).then((_) {
          sendMessage(jsonEncode({
            "sdp": peerConnection!.localDescription!.sdp!,
            "type": peerConnection!.localDescription!.type!,
          }));
        });
      } else if (type == 'answer') {
        peerConnection!.setRemoteDescription(
            RTCSessionDescription(data['sdp']!, data['type']!));
      } else if (type == 'candidate') {
        RTCIceCandidate candidate = RTCIceCandidate.fromMap(data['candidate']);
        peerConnection!.addIceCandidate(candidate);
      }
    });
  }

  Future<void> sendMessage(String message) async {
    channel!.sink.add(message);
  }

  Future<void> startLocalStream() async {
    var mediaDevices = getMediaDevices();
    var stream = await mediaDevices.getUserMedia({'video': true, 'audio': true});
    localRenderer!.srcObject = stream;
    peerConnection!.addStream(stream);
    peerConnection!.createOffer().then((RTCSessionDescription offer) {
      return peerConnection!.setLocalDescription(offer);
    }).then((_) {
      sendMessage(jsonEncode({
        "sdp": peerConnection!.localDescription!.sdp!,
        "type": peerConnection!.localDescription!.type!,
      }));
    });
  }

  @override
  Widget build(BuildContext context) {
    return Scaffold(
      appBar: AppBar(
        title: Text('WebRTC Demo'),
      ),
      body: Column(
        children: [
          Expanded(
            child: RTCVideoView(localRenderer!),
          ),
          Expanded(
            child: RTCVideoView(remoteRenderer!),
          ),
          ElevatedButton(
            onPressed: () async {
              await startLocalStream();
            },
            child: Text('Start Call'),
          ),
        ],
      ),
    );
  }

  @override
  void dispose() {
    localRenderer?.dispose();
    remoteRenderer?.dispose();
    peerConnection?.close();
    channel?.sink.close();
    super.dispose();
  }
}

注意事项

  1. 信令服务器:你需要一个信令服务器来交换SDP和ICE候选信息。上面的代码假设你有一个WebSocket信令服务器运行在wss://your-signaling-server-url
  2. 权限:确保你的应用有访问摄像头和麦克风的权限。
  3. 依赖版本:检查并更新media_flutter_webrtcweb_socket_channel的依赖版本。
  4. 错误处理:在实际应用中,你应该添加更多的错误处理逻辑来确保应用的健壮性。

这个示例代码展示了如何使用media_flutter_webrtc插件在Flutter应用中实现基本的WebRTC通信。你可以根据具体需求进行扩展和修改。

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