Flutter WebRTC视频流插件flutter_kinesis_video_webrtc的使用
Flutter WebRTC视频流插件flutter_kinesis_video_webrtc的使用
使用方法
flutter_kinesis_video_webrtc 是一个用于在Flutter应用中实现WebRTC视频流的插件，支持与AWS Kinesis Video Streams集成。以下是该插件的基本使用方法和一个完整的示例demo。
初始化PeerConnection
peerConnection() async {
// 创建SignalingClient实例，用于与Kinesis Video Streams进行信令通信
SignalingClient _signalingClient = SignalingClient(
accessKey: 'YOUR_AWS_ACCESS_KEY', // AWS访问密钥
secretKey: 'YOUR_AWS_SECRET_KEY', // AWS秘密密钥
region: 'YOUR_REGION', // AWS区域
channelName: 'YOUR_CHANNEL_NAME', // Kinesis Video Stream通道名称
);
// 初始化SignalingClient
await _signalingClient.init();
}
iOS配置
为了使应用能够访问摄像头和麦克风，需要在Info.plist文件中添加以下权限：
<key>NSCameraUsageDescription</key>
<string>$(PRODUCT_NAME) Camera Usage!</string>
<key>NSMicrophoneUsageDescription</key>
<string>$(PRODUCT_NAME) Microphone Usage!</string>
Android配置
确保在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.CAMERA" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
<uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
如果需要使用蓝牙设备，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" />
此外，建议将minSdkVersion提高到23或更高，以确保兼容性。
完整示例代码
以下是一个完整的示例代码，展示了如何在Flutter应用中使用flutter_kinesis_video_webrtc插件进行WebRTC视频流传输。
import 'dart:convert';
import 'package:flutter/foundation.dart';
import 'package:flutter/material.dart';
import 'package:flutter_kinesis_video_webrtc/flutter_kinesis_video_webrtc.dart';
import 'package:flutter_webrtc/flutter_webrtc.dart';
void main() {
runApp(const MyApp());
}
class MyApp extends StatelessWidget {
const MyApp({super.key});
[@override](/user/override)
Widget build(BuildContext context) {
return MaterialApp(
title: 'KVSWebrtcExample',
theme: ThemeData(
primarySwatch: Colors.blue,
),
home: const FlutterKinesisVideoWebrtcExample(),
);
}
}
class FlutterKinesisVideoWebrtcExample extends StatefulWidget {
const FlutterKinesisVideoWebrtcExample({super.key});
[@override](/user/override)
State<FlutterKinesisVideoWebrtcExample> createState() => _FlutterKinesisVideoWebrtcExampleState();
}
class _FlutterKinesisVideoWebrtcExampleState extends State<FlutterKinesisVideoWebrtcExample> {
final TextEditingController _accessKeyController = TextEditingController();
final TextEditingController _secretKeyController = TextEditingController();
final TextEditingController _regionController = TextEditingController();
final TextEditingController _channelNameController = TextEditingController();
final RTCVideoRenderer _rtcVideoRenderer = RTCVideoRenderer();
RTCPeerConnection? _rtcPeerConnection;
late SignalingClient _signalingClient;
bool sendAudio = false;
bool sendVideo = false;
MediaStream? _localStream;
[@override](/user/override)
void initState() {
// 初始化RTCVideoRenderer
_rtcVideoRenderer.initialize();
super.initState();
}
[@override](/user/override)
void dispose() {
// 释放资源
_accessKeyController.dispose();
_secretKeyController.dispose();
_regionController.dispose();
_channelNameController.dispose();
_rtcVideoRenderer.dispose();
_rtcPeerConnection?.dispose();
super.dispose();
}
peerConnection() async {
// 创建SignalingClient实例
_signalingClient = SignalingClient(
channelName: _channelNameController.text.trim(),
accessKey: _accessKeyController.text.trim(),
secretKey: _secretKeyController.text.trim(),
region: _regionController.text.trim(),
);
// 初始化SignalingClient
await _signalingClient.init();
// 创建RTCPeerConnection
_rtcPeerConnection = await createPeerConnection({
'iceServers': _signalingClient.iceServers,
'iceTransportPolicy': 'all'
});
// 处理接收到的视频流
_rtcPeerConnection!.onTrack = (event) {
_rtcVideoRenderer.srcObject = event.streams[0];
setState(() {});
};
// 如果需要发送音频和视频
if (sendAudio || sendVideo) {
_localStream = await navigator.mediaDevices.getUserMedia({
'audio': sendAudio,
'video': sendVideo,
});
_localStream!.getTracks().forEach((track) {
_rtcPeerConnection!.addTrack(track, _localStream!);
setState(() {});
});
}
// 创建WebSocket连接
var webSocket = SimpleWebSocket(_signalingClient.domain ?? '',
_signalingClient.signedQueryParams ?? <String, dynamic>{});
// 处理接收到的消息
webSocket.onMessage = (data) async {
if (data != '') {
var objectOfData = jsonDecode(data);
print("-------------------- receiving ${objectOfData['messageType']} --------------------");
if (objectOfData['messageType'] == "SDP_ANSWER") {
var decodedAns = jsonDecode(utf8.decode(base64.decode(objectOfData['messagePayload'])));
await _rtcPeerConnection?.setRemoteDescription(RTCSessionDescription(
decodedAns["sdp"],
decodedAns["type"],
));
} else if (objectOfData['messageType'] == "ICE_CANDIDATE") {
var decodedCandidate = jsonDecode(utf8.decode(base64.decode(objectOfData['messagePayload'])));
await _rtcPeerConnection?.addCandidate(
RTCIceCandidate(decodedCandidate["candidate"], decodedCandidate["sdpMid"], decodedCandidate["sdpMLineIndex"]),
);
}
}
};
// WebSocket连接打开时发送SDP_OFFER
webSocket.onOpen = () async {
if (kDebugMode) {
print("-------------------- socket opened --------------------");
print("-------------------- sending 'SDP_OFFER' --------------------");
}
RTCSessionDescription offer = await _rtcPeerConnection!.createOffer({
'mandatory': {
'OfferToReceiveAudio': true,
'OfferToReceiveVideo': true,
},
'optional': [],
});
await _rtcPeerConnection!.setLocalDescription(offer);
RTCSessionDescription? localDescription = await _rtcPeerConnection?.getLocalDescription();
var request = {};
request["action"] = "SDP_OFFER";
request["messagePayload"] = base64.encode(jsonEncode(localDescription?.toMap()).codeUnits);
webSocket.send(jsonEncode(request));
};
// 处理ICE候选
_rtcPeerConnection!.onIceCandidate = (dynamic candidate) {
if (kDebugMode) {
print("-------------------- sending 'ICE_CANDIDATE' --------------------");
}
var request = {};
request["action"] = "ICE_CANDIDATE";
request["messagePayload"] = base64.encode(jsonEncode(candidate.toMap()).codeUnits);
webSocket.send(jsonEncode(request));
};
// 连接WebSocket
await webSocket.connect();
}
[@override](/user/override)
Widget build(BuildContext context) {
return Scaffold(
appBar: AppBar(
title: const Text('Example'),
),
body: SafeArea(
child: SingleChildScrollView(
child: SizedBox(
height: MediaQuery.of(context).size.height * 1.4,
child: Column(
children: [
Expanded(
flex: 4,
child: Padding(
padding: const EdgeInsets.all(20),
child: Column(
crossAxisAlignment: CrossAxisAlignment.start,
children: [
const Text("ACCESS-KEY", textScaleFactor: 1.0),
TextFormField(
keyboardType: TextInputType.text,
maxLength: 128,
controller: _accessKeyController,
decoration: const InputDecoration(
hintText: "Enter access key",
counterText: "",
),
),
const SizedBox(height: 20),
const Text("SECRET-KEY", textScaleFactor: 1.0),
TextFormField(
keyboardType: TextInputType.visiblePassword,
obscureText: true,
controller: _secretKeyController,
decoration: const InputDecoration(
hintText: "Enter secret key",
counterText: "",
),
),
const SizedBox(height: 20),
const Text("REGION", textScaleFactor: 1.0),
TextFormField(
keyboardType: TextInputType.text,
controller: _regionController,
decoration: const InputDecoration(
hintText: "Enter region ex: us-east-1",
),
),
const SizedBox(height: 20),
const Text("CHANNEL NAME", textScaleFactor: 1.0),
TextFormField(
keyboardType: TextInputType.text,
controller: _channelNameController,
decoration: const InputDecoration(
hintText: "Enter channel name",
),
),
const SizedBox(height: 20),
Center(
child: ElevatedButton(
onPressed: () {
peerConnection();
setState(() {});
},
child: Container(
padding: const EdgeInsets.symmetric(vertical: 20),
width: MediaQuery.of(context).size.width - 80,
child: const Text(
"START WEBRTC",
textAlign: TextAlign.center,
textScaleFactor: 1.0,
),
),
),
),
],
),
),
),
Expanded(
flex: 6,
child: _rtcVideoRenderer.renderVideo
? Center(
child: AspectRatio(
aspectRatio: _rtcVideoRenderer.value.aspectRatio,
child: RTCVideoView(_rtcVideoRenderer),
),
)
: const Center(
child: CircularProgressIndicator(),
),
),
],
),
),
),
),
);
}
}
更多关于Flutter WebRTC视频流插件flutter_kinesis_video_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视频流插件flutter_kinesis_video_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，关于如何在Flutter项目中使用flutter_kinesis_video_webrtc插件来实现WebRTC视频流，以下是一个简单的代码示例。这个示例展示了如何设置和连接WebRTC客户端。请注意，flutter_kinesis_video_webrtc插件主要用于与Amazon Kinesis Video Streams进行WebRTC通信，所以你可能需要有一个Kinesis Video Stream来测试这个示例。
1. 添加依赖
首先，在你的pubspec.yaml文件中添加flutter_kinesis_video_webrtc依赖：
dependencies:
flutter:
sdk: flutter
flutter_kinesis_video_webrtc: ^x.y.z # 替换为最新版本号
然后运行flutter pub get来安装依赖。
2. 配置权限（Android）
在android/app/src/main/AndroidManifest.xml中添加必要的权限：
<uses-permission android:name="android.permission.INTERNET"/>
<uses-permission android:name="android.permission.CAMERA"/>
<uses-permission android:name="android.permission.RECORD_AUDIO"/>
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS"/>
3. 实现WebRTC连接
以下是一个简单的Flutter应用示例，展示了如何使用flutter_kinesis_video_webrtc进行WebRTC连接：
import 'package:flutter/material.dart';
import 'package:flutter_kinesis_video_webrtc/flutter_kinesis_video_webrtc.dart';
void main() {
runApp(MyApp());
}
class MyApp extends StatefulWidget {
@override
_MyAppState createState() => _MyAppState();
}
class _MyAppState extends State<MyApp> {
KinesisVideoWebRTCClient? _client;
String _signalingUrl = 'YOUR_SIGNALING_SERVER_URL'; // 替换为你的信令服务器URL
@override
void initState() {
super.initState();
initWebRTC();
}
Future<void> initWebRTC() async {
_client = KinesisVideoWebRTCClient(
signalingUrl: _signalingUrl,
iceServers: [
// 添加你的ICE服务器配置
IceServer('stun:stun.l.google.com:19302'),
],
);
_client!.onSignalingChange = (SignalingState state) {
print('Signaling state changed: $state');
if (state == SignalingState.CONNECTED) {
// 连接成功后可以开始创建Offer或Answer
createOffer();
}
};
_client!.onRemoteStreamAdded = (MediaStream stream) {
print('Remote stream added');
// 在这里处理远程视频流，比如将其显示在一个VideoPlayer上
};
await _client!.connect();
}
Future<void> createOffer() async {
if (_client!.signalingState == SignalingState.CONNECTED) {
RTCSessionDescription? offer = await _client!.createOffer();
await _client!.setLocalDescription(offer!);
// 发送Offer到你的信令服务器
// await signalingServer.sendOffer(offer.sdp);
}
}
@override
Widget build(BuildContext context) {
return MaterialApp(
home: Scaffold(
appBar: AppBar(
title: Text('Flutter Kinesis Video WebRTC Example'),
),
body: Center(
child: Text('Connecting to WebRTC...'),
),
),
);
}
@override
void dispose() {
_client?.close();
super.dispose();
}
}
注意事项
-
信令服务器：这个示例假设你有一个信令服务器来处理WebRTC的信令交换。你需要将
_signalingUrl替换为你的信令服务器URL，并实现与信令服务器的通信（例如发送Offer和Answer）。 -
ICE服务器：示例中使用了Google的STUN服务器。如果你需要TURN服务器来处理NAT穿越，你需要在
iceServers列表中添加相应的配置。 -
错误处理：为了简洁，示例中没有包含错误处理逻辑。在实际应用中，你应该添加适当的错误处理来确保应用的健壮性。
-
UI更新：示例中的UI非常简单，只显示了一个连接中的文本。在实际应用中，你可能需要更新UI以显示本地和远程视频流，以及处理各种WebRTC事件（如连接状态变化、流添加/移除等）。
这个示例提供了一个基本的框架，你可以根据具体需求进行扩展和修改。
