Flutter WebRTC通信插件medea_flutter_webrtc的使用
Flutter WebRTC通信插件 medea_flutter_webrtc
的使用
简介
medea_flutter_webrtc
是一个用于 Flutter 的 WebRTC 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
在 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
在 <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
在 <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
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
- 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
。 - 权限:确保你的应用有访问摄像头和麦克风的权限。
- 依赖版本:检查并更新
media_flutter_webrtc
和web_socket_channel
的依赖版本。 - 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。