16 changed files with 3252 additions and 0 deletions
@ -0,0 +1,51 @@ |
|||
# Miscellaneous |
|||
*.class |
|||
*.log |
|||
*.pyc |
|||
*.swp |
|||
.DS_Store |
|||
.atom/ |
|||
.buildlog/ |
|||
.history |
|||
.svn/ |
|||
|
|||
# IntelliJ related |
|||
*.iml |
|||
*.ipr |
|||
*.iws |
|||
.idea/ |
|||
|
|||
# Visual Studio Code related |
|||
.vscode/ |
|||
|
|||
# Flutter/Dart/Pub related |
|||
**/doc/api/ |
|||
.dart_tool/ |
|||
.flutter-plugins |
|||
.flutter-plugins-dependencies |
|||
.packages |
|||
.pub-cache/ |
|||
.pub/ |
|||
build/ |
|||
|
|||
# iOS related |
|||
**/ios/.generated/ |
|||
**/ios/Flutter/Flutter.framework |
|||
**/ios/Flutter/Flutter.podspec |
|||
**/ios/Flutter/Generated.xcconfig |
|||
**/ios/Flutter/app.flx |
|||
**/ios/Flutter/app.zip |
|||
**/ios/Flutter/flutter_assets/ |
|||
**/ios/Flutter/flutter_export_environment.sh |
|||
**/ios/ServiceDefinitions.json |
|||
**/ios/Runner/GeneratedPluginRegistrant.* |
|||
.build/ |
|||
|
|||
# Coverage |
|||
coverage/ |
|||
|
|||
# Exceptions to above rules. |
|||
!**/ios/**/default.mode1v3 |
|||
!**/ios/**/default.mode2v3 |
|||
!**/ios/**/default.pbxuser |
|||
!**/ios/**/default.perspectivev3 |
|||
@ -0,0 +1,190 @@ |
|||
# Realtime Plugin 实现总结 |
|||
|
|||
## 项目概述 |
|||
|
|||
基于您提供的Android原生示例,我为DeepVoice项目创建了一个realtime插件,用于通过WebSocket接入Vocode服务器实现实时语音聊天功能。该插件已完成Android和iOS双平台实现。 |
|||
|
|||
## 架构设计 |
|||
|
|||
### 插件结构 |
|||
``` |
|||
local_plugins/realtime/ |
|||
├── android/ # Android实现 |
|||
│ ├── build.gradle # Android构建配置 |
|||
│ ├── src/main/ |
|||
│ │ ├── AndroidManifest.xml # Android权限配置 |
|||
│ │ └── kotlin/com/yunqiinnovation/realtime/ |
|||
│ │ ├── RealtimePlugin.kt # 主插件入口 |
|||
│ │ ├── RealtimeAudioManager.kt # 音频管理器(AudioRecord/AudioTrack) |
|||
│ │ └── RealtimeWebSocketManager.kt # WebSocket管理器(OkHttp) |
|||
├── ios/ # iOS实现 |
|||
│ ├── Classes/ # Objective-C桥接文件 |
|||
│ │ ├── RealtimePlugin.h |
|||
│ │ └── RealtimePlugin.m |
|||
│ ├── realtime/ # Swift Package |
|||
│ │ ├── Package.swift |
|||
│ │ └── Sources/realtime/ |
|||
│ │ ├── RealtimePlugin.swift # 主插件入口 |
|||
│ │ ├── RealtimeAudioManager.swift # 音频管理器(AVAudioEngine) |
|||
│ │ └── RealtimeWebSocketManager.swift # WebSocket管理器(URLSession) |
|||
│ └── realtime.podspec |
|||
├── lib/ |
|||
│ └── realtime.dart # Flutter接口 |
|||
├── pubspec.yaml |
|||
└── README.md |
|||
``` |
|||
|
|||
### 核心组件 |
|||
|
|||
1. **RealtimePlugin (SwiftRealtimePlugin)** |
|||
- 主插件入口,处理Flutter方法调用 |
|||
- 管理各组件间的协调 |
|||
- 处理事件分发 |
|||
|
|||
2. **RealtimeAudioManager** |
|||
- 使用AVAudioEngine进行音频录制 |
|||
- 使用AVAudioPlayerNode进行音频播放 |
|||
- 支持16kHz/16-bit/单声道格式 |
|||
- 实现20ms帧长的音频处理 |
|||
|
|||
3. **RealtimeWebSocketManager** |
|||
- 使用URLSessionWebSocketTask实现WebSocket通信 |
|||
- 支持音频数据和文本消息的收发 |
|||
- 自动连接状态管理 |
|||
|
|||
## 技术实现对照 |
|||
|
|||
| 功能 | Android示例 | iOS实现 | |
|||
|------|-------------|---------| |
|||
| 音频录制 | AudioRecord | AVAudioEngine + AVAudioInputNode | |
|||
| 音频播放 | AudioTrack | AVAudioPlayerNode | |
|||
| WebSocket | OkHttp WebSocket | URLSessionWebSocketTask | |
|||
| 线程管理 | Kotlin协程 | DispatchQueue | |
|||
| 音频格式 | 16kHz/16-bit/单声道 | 16kHz/16-bit/单声道 | |
|||
| 帧长 | 20ms (320字节) | 20ms (320字节) | |
|||
|
|||
## Flutter接口 |
|||
|
|||
### 主要类 |
|||
- `RealtimeService`: 主服务类 |
|||
- `RealtimeEvent`: 事件类 |
|||
- `ConnectionStatus`: 连接状态枚举 |
|||
- `VoiceStatus`: 语音状态枚举 |
|||
- `RealtimeException`: 异常类 |
|||
|
|||
### 主要方法 |
|||
- `initialize()`: 初始化服务 |
|||
- `connect()`: 连接服务器 |
|||
- `disconnect()`: 断开连接 |
|||
- `startRecording()`: 开始录音 |
|||
- `stopRecording()`: 停止录音 |
|||
- `eventStream`: 事件流 |
|||
|
|||
## 使用方式 |
|||
|
|||
### 1. 在RealtimeController中集成 |
|||
已更新`lib/modules/realtime/controllers/realtime_controller.dart`使用真实的realtime插件,取代了原来的模拟实现。 |
|||
|
|||
### 2. 事件监听 |
|||
```dart |
|||
_realtimeService.eventStream.listen((event) { |
|||
switch (event.type) { |
|||
case 'connectionStatusChanged': |
|||
// 处理连接状态变化 |
|||
case 'voiceStatusChanged': |
|||
// 处理语音状态变化 |
|||
case 'textReceived': |
|||
// 处理文本消息 |
|||
case 'error': |
|||
// 处理错误 |
|||
} |
|||
}); |
|||
``` |
|||
|
|||
## 配置要求 |
|||
|
|||
### 依赖配置 |
|||
已添加到主项目的`pubspec.yaml`: |
|||
```yaml |
|||
realtime: |
|||
path: local_plugins/realtime |
|||
``` |
|||
|
|||
### 权限配置 |
|||
iOS的`Info.plist`已包含必要的麦克风权限: |
|||
```xml |
|||
<key>NSMicrophoneUsageDescription</key> |
|||
<string>需要麦克风权限用于语音识别和录音功能</string> |
|||
``` |
|||
|
|||
## 与Android示例的对应关系 |
|||
|
|||
1. **初始化对应** |
|||
- Android: 创建AudioRecord, AudioTrack, OkHttpClient |
|||
- iOS: 创建AVAudioEngine, AVAudioPlayerNode, URLSession |
|||
|
|||
2. **录音线程对应** |
|||
- Android: `loopRecordSend()` 协程 |
|||
- iOS: AVAudioInputNode的installTap回调 |
|||
|
|||
3. **播放线程对应** |
|||
- Android: `loopPlayback()` 协程 + LinkedBlockingQueue |
|||
- iOS: DispatchQueue + AVAudioPCMBuffer队列 |
|||
|
|||
4. **WebSocket对应** |
|||
- Android: OkHttp WebSocketListener |
|||
- iOS: URLSessionWebSocketDelegate |
|||
|
|||
## 特性支持 |
|||
|
|||
✅ **已实现** |
|||
- Android和iOS双平台支持 |
|||
- 实时音频录制和播放(16kHz/16-bit/单声道) |
|||
- WebSocket双向通信 |
|||
- 状态管理和事件通知 |
|||
- 错误处理 |
|||
- 资源管理和清理 |
|||
- 20ms帧长处理 |
|||
- Kotlin协程和DispatchQueue线程管理 |
|||
|
|||
❌ **未实现(可扩展)** |
|||
- 音频编码(Opus等) |
|||
- 噪声消除/回声抑制 |
|||
- 自动重连机制 |
|||
- 音频质量自适应 |
|||
|
|||
## 使用注意事项 |
|||
|
|||
1. **服务器地址配置** |
|||
```dart |
|||
// 需要替换为实际的Vocode服务器地址 |
|||
serverUrl: 'wss://your-vocode-server/ws' |
|||
``` |
|||
|
|||
2. **音频格式一致性** |
|||
- 确保服务器支持16kHz/16-bit/单声道格式 |
|||
- 帧长固定为20ms (320字节) |
|||
|
|||
3. **错误处理** |
|||
- 监听事件流中的错误 |
|||
- 处理网络断线和重连 |
|||
|
|||
4. **资源管理** |
|||
- 及时调用dispose()释放资源 |
|||
- 避免内存泄漏 |
|||
|
|||
## 测试建议 |
|||
|
|||
1. **本地测试** |
|||
- 先用echo服务器测试WebSocket连接 |
|||
- 验证音频录制和播放功能 |
|||
|
|||
2. **集成测试** |
|||
- 与真实Vocode服务器集成 |
|||
- 测试端到端语音交互 |
|||
|
|||
3. **性能测试** |
|||
- 测试长时间使用的稳定性 |
|||
- 监控内存和CPU使用 |
|||
|
|||
这个实现为DeepVoice项目提供了完整的实时语音交互能力,可以直接与Vocode服务器进行通信,实现类似Android示例的功能。 |
|||
@ -0,0 +1,125 @@ |
|||
# Realtime Plugin |
|||
|
|||
实时语音聊天插件,通过WebSocket连接Vocode服务器实现实时语音交互。 |
|||
|
|||
## 功能特性 |
|||
|
|||
- **实时音频录制**: 16kHz/16-bit/单声道格式录音 |
|||
- **WebSocket通信**: 与Vocode服务器进行实时数据传输 |
|||
- **音频播放**: 播放服务器返回的语音数据 |
|||
- **状态管理**: 连接状态和语音状态监控 |
|||
- **事件回调**: 支持各种事件的监听和处理 |
|||
|
|||
## 技术实现 |
|||
|
|||
### Android端实现 |
|||
- **音频录制**: 使用AudioRecord进行16kHz/16-bit/单声道录音 |
|||
- **音频播放**: 使用AudioTrack进行音频播放 |
|||
- **WebSocket**: 使用OkHttp WebSocket客户端 |
|||
- **线程管理**: 使用Kotlin协程处理录音和播放 |
|||
|
|||
### iOS端实现 |
|||
- **音频录制**: 使用AVAudioEngine和AVAudioInputNode |
|||
- **音频播放**: 使用AVAudioPlayerNode进行播放 |
|||
- **WebSocket**: 使用URLSessionWebSocketTask |
|||
- **线程管理**: 使用DispatchQueue管理录音和播放线程 |
|||
|
|||
### 核心组件 |
|||
- `RealtimePlugin`: 主插件入口,处理Flutter方法调用 |
|||
- `RealtimeAudioManager`: 音频管理器,负责录音和播放 |
|||
- `RealtimeWebSocketManager`: WebSocket管理器,负责网络通信 |
|||
|
|||
**Android端组件**: |
|||
- AudioRecord + AudioTrack + OkHttp + Kotlin协程 |
|||
- 对应您提供的Android原生示例功能 |
|||
|
|||
**iOS端组件**: |
|||
- AVAudioEngine + AVAudioPlayerNode + URLSessionWebSocketTask + DispatchQueue |
|||
|
|||
## 使用方法 |
|||
|
|||
```dart |
|||
import 'package:realtime/realtime.dart'; |
|||
|
|||
final realtimeService = RealtimeService(); |
|||
|
|||
// 初始化 |
|||
await realtimeService.initialize( |
|||
serverUrl: 'wss://your-server/ws', |
|||
sampleRate: 16000, |
|||
channels: 1, |
|||
bitsPerSample: 16, |
|||
); |
|||
|
|||
// 监听事件 |
|||
realtimeService.eventStream.listen((event) { |
|||
switch (event.type) { |
|||
case 'connectionStatusChanged': |
|||
// 处理连接状态变化 |
|||
break; |
|||
case 'voiceStatusChanged': |
|||
// 处理语音状态变化 |
|||
break; |
|||
case 'textReceived': |
|||
// 处理收到的文本消息 |
|||
break; |
|||
case 'error': |
|||
// 处理错误 |
|||
break; |
|||
} |
|||
}); |
|||
|
|||
// 连接服务器 |
|||
await realtimeService.connect(); |
|||
|
|||
// 开始录音 |
|||
await realtimeService.startRecording(); |
|||
|
|||
// 停止录音 |
|||
await realtimeService.stopRecording(); |
|||
|
|||
// 断开连接 |
|||
await realtimeService.disconnect(); |
|||
|
|||
// 释放资源 |
|||
await realtimeService.dispose(); |
|||
``` |
|||
|
|||
## 音频格式 |
|||
|
|||
- **采样率**: 16000 Hz |
|||
- **声道数**: 1(单声道) |
|||
- **位深**: 16-bit |
|||
- **帧长**: 20ms (320字节) |
|||
|
|||
## 权限要求 |
|||
|
|||
### Android |
|||
权限已自动包含在插件中: |
|||
- `RECORD_AUDIO`: 音频录制权限 |
|||
- `INTERNET`: 网络访问权限 |
|||
- `ACCESS_NETWORK_STATE`: 网络状态权限 |
|||
|
|||
### iOS |
|||
在Info.plist中添加麦克风权限: |
|||
```xml |
|||
<key>NSMicrophoneUsageDescription</key> |
|||
<string>应用需要麦克风权限进行语音录制</string> |
|||
``` |
|||
|
|||
## 注意事项 |
|||
|
|||
1. 确保服务器地址正确且可访问 |
|||
2. 网络环境良好,避免频繁断线 |
|||
3. 音频格式与服务器保持一致 |
|||
4. 及时释放资源,避免内存泄漏 |
|||
|
|||
## 错误处理 |
|||
|
|||
插件会自动处理常见错误: |
|||
- 网络连接失败 |
|||
- 音频设备不可用 |
|||
- 权限被拒绝 |
|||
- 服务器断开连接 |
|||
|
|||
通过事件流可以监听这些错误并进行相应处理。 |
|||
@ -0,0 +1,47 @@ |
|||
plugins { |
|||
id("com.android.library") |
|||
id("org.jetbrains.kotlin.android") |
|||
kotlin("plugin.serialization") version "1.9.24" |
|||
} |
|||
|
|||
android { |
|||
namespace = "com.yunqiinnovation.realtime" |
|||
compileSdk = 35 |
|||
|
|||
defaultConfig { |
|||
minSdk = 21 |
|||
targetSdk = 33 |
|||
} |
|||
|
|||
compileOptions { |
|||
sourceCompatibility = JavaVersion.VERSION_11 |
|||
targetCompatibility = JavaVersion.VERSION_11 |
|||
} |
|||
|
|||
kotlinOptions { |
|||
jvmTarget = "11" |
|||
} |
|||
|
|||
packaging { |
|||
resources { |
|||
excludes.add("META-INF/DEPENDENCIES") |
|||
excludes.add("META-INF/LICENSE") |
|||
excludes.add("META-INF/LICENSE.txt") |
|||
excludes.add("META-INF/license.txt") |
|||
excludes.add("META-INF/NOTICE") |
|||
excludes.add("META-INF/NOTICE.txt") |
|||
excludes.add("META-INF/notice.txt") |
|||
excludes.add("META-INF/*.kotlin_module") |
|||
} |
|||
} |
|||
} |
|||
|
|||
dependencies { |
|||
implementation("org.jetbrains.kotlin:kotlin-stdlib-jdk8:1.9.10") |
|||
implementation("org.jetbrains.kotlinx:kotlinx-coroutines-core:1.7.3") |
|||
implementation("org.jetbrains.kotlinx:kotlinx-coroutines-android:1.7.3") |
|||
|
|||
// OkHttp for WebSocket |
|||
implementation(platform("com.squareup.okhttp3:okhttp-bom:4.12.0")) |
|||
implementation("com.squareup.okhttp3:okhttp") |
|||
} |
|||
@ -0,0 +1 @@ |
|||
rootProject.name = "realtime" |
|||
@ -0,0 +1,11 @@ |
|||
<manifest xmlns:android="http://schemas.android.com/apk/res/android" |
|||
package="com.yunqiinnovation.realtime"> |
|||
|
|||
<!-- 音频录制权限 --> |
|||
<uses-permission android:name="android.permission.RECORD_AUDIO" /> |
|||
<!-- 网络权限 --> |
|||
<uses-permission android:name="android.permission.INTERNET" /> |
|||
<!-- 网络状态权限 --> |
|||
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" /> |
|||
|
|||
</manifest> |
|||
@ -0,0 +1,519 @@ |
|||
package com.yunqiinnovation.realtime |
|||
|
|||
import android.content.Context |
|||
import android.media.* |
|||
import android.media.audiofx.AcousticEchoCanceler |
|||
import android.media.audiofx.AutomaticGainControl |
|||
import android.media.audiofx.NoiseSuppressor |
|||
import android.util.Log |
|||
import kotlinx.coroutines.* |
|||
import java.io.ByteArrayOutputStream |
|||
import java.nio.ByteBuffer |
|||
import java.nio.ByteOrder |
|||
import java.util.concurrent.LinkedBlockingQueue |
|||
import kotlin.math.sqrt |
|||
|
|||
|
|||
/** |
|||
* 实时音频管理器 |
|||
* |
|||
* 负责音频的录制和播放,对应Android示例中的AudioRecord和AudioTrack功能 |
|||
*/ |
|||
class RealtimeAudioManager(private val context: Context) { |
|||
companion object { |
|||
private const val TAG = "RealtimeAudioManager" |
|||
} |
|||
|
|||
// 音频配置 |
|||
private var audioConfig: AudioConfig? = null |
|||
|
|||
// 录音相关 |
|||
private var audioRecord: AudioRecord? = null |
|||
private var isRecording = false |
|||
private var recordJob: Job? = null |
|||
|
|||
// 播放相关 |
|||
private var audioTrack: AudioTrack? = null |
|||
private var isPlaying = false |
|||
private var playJob: Job? = null |
|||
private val playbackQueue = LinkedBlockingQueue<ByteArray>() |
|||
private val playbackInternalBuffer = ByteArrayOutputStream() |
|||
|
|||
// 回音消除相关 |
|||
private var acousticEchoCanceler: AcousticEchoCanceler? = null |
|||
private var noiseSuppressor: NoiseSuppressor? = null |
|||
private var automaticGainControl: AutomaticGainControl? = null |
|||
private var audioManager: AudioManager? = null |
|||
private var originalAudioMode = AudioManager.MODE_NORMAL |
|||
|
|||
// 回调 |
|||
var onAudioData: ((ByteArray) -> Unit)? = null |
|||
var onStatusChanged: ((VoiceStatus) -> Unit)? = null |
|||
var onRmsChanged: ((Float) -> Unit)? = null // 新增RMS回调 |
|||
|
|||
// 协程作用域 |
|||
private val scope = CoroutineScope(Dispatchers.IO + SupervisorJob()) |
|||
|
|||
/// 初始化音频管理器 |
|||
fun initialize(config: AudioConfig): Boolean { |
|||
this.audioConfig = config |
|||
|
|||
// 初始化系统音频管理器 |
|||
audioManager = context.getSystemService(Context.AUDIO_SERVICE) as AudioManager |
|||
|
|||
// 保存原始音频模式 |
|||
originalAudioMode = audioManager?.mode ?: AudioManager.MODE_NORMAL |
|||
|
|||
return setupAudioRecord(config) && setupAudioTrack(config) |
|||
} |
|||
|
|||
/// 设置AudioRecord(录音) |
|||
private fun setupAudioRecord(config: AudioConfig): Boolean { |
|||
try { |
|||
val channelConfig = if (config.channels == 1) { |
|||
AudioFormat.CHANNEL_IN_MONO |
|||
} else { |
|||
AudioFormat.CHANNEL_IN_STEREO |
|||
} |
|||
|
|||
val audioFormat = when (config.bitsPerSample) { |
|||
16 -> AudioFormat.ENCODING_PCM_16BIT |
|||
8 -> AudioFormat.ENCODING_PCM_8BIT |
|||
else -> AudioFormat.ENCODING_PCM_16BIT |
|||
} |
|||
|
|||
val bufferSize = AudioRecord.getMinBufferSize( |
|||
config.sampleRate, |
|||
channelConfig, |
|||
audioFormat |
|||
) |
|||
|
|||
if (bufferSize == AudioRecord.ERROR || bufferSize == AudioRecord.ERROR_BAD_VALUE) { |
|||
Log.e(TAG, "无法获取音频录制缓冲区大小") |
|||
return false |
|||
} |
|||
|
|||
audioRecord = AudioRecord( |
|||
MediaRecorder.AudioSource.MIC, |
|||
config.sampleRate, |
|||
channelConfig, |
|||
audioFormat, |
|||
bufferSize |
|||
) |
|||
|
|||
if (audioRecord?.state != AudioRecord.STATE_INITIALIZED) { |
|||
Log.e(TAG, "AudioRecord初始化失败") |
|||
return false |
|||
} |
|||
|
|||
Log.i(TAG, "AudioRecord初始化成功,采样率: ${config.sampleRate}, 声道: ${config.channels}, 位深: ${config.bitsPerSample}") |
|||
return true |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "设置AudioRecord失败: ${e.message}", e) |
|||
return false |
|||
} |
|||
} |
|||
|
|||
/// 设置AudioTrack(播放) |
|||
private fun setupAudioTrack(config: AudioConfig): Boolean { |
|||
try { |
|||
val channelConfig = if (config.channels == 1) { |
|||
AudioFormat.CHANNEL_OUT_MONO |
|||
} else { |
|||
AudioFormat.CHANNEL_OUT_STEREO |
|||
} |
|||
|
|||
val audioFormat = when (config.bitsPerSample) { |
|||
16 -> AudioFormat.ENCODING_PCM_16BIT |
|||
8 -> AudioFormat.ENCODING_PCM_8BIT |
|||
else -> AudioFormat.ENCODING_PCM_16BIT |
|||
} |
|||
|
|||
val bufferSize = AudioTrack.getMinBufferSize( |
|||
config.sampleRate, |
|||
channelConfig, |
|||
audioFormat |
|||
) |
|||
|
|||
if (bufferSize == AudioTrack.ERROR || bufferSize == AudioTrack.ERROR_BAD_VALUE) { |
|||
Log.e(TAG, "无法获取音频播放缓冲区大小") |
|||
return false |
|||
} |
|||
|
|||
// 总是使用VOICE_COMMUNICATION模式以获得最佳语音处理效果 |
|||
val audioAttributes = AudioAttributes.Builder() |
|||
.setUsage(AudioAttributes.USAGE_VOICE_COMMUNICATION) |
|||
.setContentType(AudioAttributes.CONTENT_TYPE_SPEECH) |
|||
.build() |
|||
|
|||
audioTrack = AudioTrack.Builder() |
|||
.setAudioAttributes(audioAttributes) |
|||
.setAudioFormat( |
|||
AudioFormat.Builder() |
|||
.setSampleRate(config.sampleRate) |
|||
.setChannelMask(channelConfig) |
|||
.setEncoding(audioFormat) |
|||
.build() |
|||
) |
|||
.setBufferSizeInBytes(bufferSize) |
|||
.setTransferMode(AudioTrack.MODE_STREAM) |
|||
.build() |
|||
|
|||
if (audioTrack?.state != AudioTrack.STATE_INITIALIZED) { |
|||
Log.e(TAG, "AudioTrack初始化失败") |
|||
return false |
|||
} |
|||
|
|||
Log.i(TAG, "AudioTrack初始化成功, 启用语音通信模式") |
|||
return true |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "设置AudioTrack失败: ${e.message}", e) |
|||
return false |
|||
} |
|||
} |
|||
|
|||
/// 开始录音 |
|||
fun startRecording(): Boolean { |
|||
if (isRecording || audioRecord == null) { |
|||
return false |
|||
} |
|||
|
|||
try { |
|||
// 设置回音消除(总是启用) |
|||
setupEchoCancellation() |
|||
|
|||
audioRecord?.startRecording() |
|||
isRecording = true |
|||
onStatusChanged?.invoke(VoiceStatus.RECORDING) |
|||
|
|||
// 启动录音协程 |
|||
recordJob = scope.launch { |
|||
loopRecordSend() |
|||
} |
|||
|
|||
Log.i(TAG, "开始录音, 回音消除: ${acousticEchoCanceler?.enabled}") |
|||
return true |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "开始录音失败: ${e.message}", e) |
|||
return false |
|||
} |
|||
} |
|||
|
|||
/// 录音循环(对应Android示例中的loopRecordSend) |
|||
private suspend fun loopRecordSend() { |
|||
val config = audioConfig ?: return |
|||
val frameSize = config.frameSize |
|||
val buffer = ByteArray(frameSize) |
|||
|
|||
while (isRecording && audioRecord != null) { |
|||
try { |
|||
val bytesRead = audioRecord!!.read(buffer, 0, frameSize) |
|||
|
|||
if (bytesRead > 0) { |
|||
// 确保读取的数据长度正确 |
|||
val audioData = if (bytesRead == frameSize) { |
|||
buffer |
|||
} else { |
|||
buffer.copyOf(bytesRead) |
|||
} |
|||
|
|||
// 计算RMS值 |
|||
val rms = calculateRMS(audioData) |
|||
onRmsChanged?.invoke(rms) |
|||
|
|||
// 直接回调音频数据到Flutter层 |
|||
onAudioData?.invoke(audioData) |
|||
} else { |
|||
Log.w(TAG, "录音读取数据失败: $bytesRead") |
|||
} |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "录音循环异常: ${e.message}", e) |
|||
break |
|||
} |
|||
} |
|||
} |
|||
|
|||
/// 停止录音 |
|||
fun stopRecording(): Boolean { |
|||
if (!isRecording) { |
|||
return false |
|||
} |
|||
|
|||
try { |
|||
isRecording = false |
|||
recordJob?.cancel() |
|||
audioRecord?.stop() |
|||
|
|||
// 清理回音消除 |
|||
cleanupEchoCancellation() |
|||
|
|||
onStatusChanged?.invoke(VoiceStatus.IDLE) |
|||
Log.i(TAG, "录音已停止") |
|||
return true |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "停止录音失败: ${e.message}", e) |
|||
return false |
|||
} |
|||
} |
|||
|
|||
/// 播放音频数据 |
|||
fun playAudioData(data: ByteArray) { |
|||
if (data.isNotEmpty()) { |
|||
// 计算播放数据的RMS值 |
|||
val rms = calculateRMS(data) |
|||
onRmsChanged?.invoke(rms) |
|||
|
|||
playbackQueue.offer(data) |
|||
if (!isPlaying) { |
|||
startPlayback() |
|||
} |
|||
} |
|||
} |
|||
|
|||
/// 开始播放 |
|||
private fun startPlayback() { |
|||
if (isPlaying || audioTrack == null) { |
|||
return |
|||
} |
|||
|
|||
try { |
|||
isPlaying = true |
|||
onStatusChanged?.invoke(VoiceStatus.PLAYING) |
|||
audioTrack?.play() |
|||
|
|||
playJob = scope.launch { |
|||
loopPlayback() |
|||
} |
|||
Log.i(TAG, "开始播放") |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "开始播放失败: ${e.message}", e) |
|||
} |
|||
} |
|||
|
|||
/// 播放循环 |
|||
private suspend fun loopPlayback() { |
|||
Log.d(TAG, "启动播放循环") |
|||
|
|||
while (isPlaying) { |
|||
try { |
|||
// take() 会一直等待直到队列中有数据 |
|||
val data = playbackQueue.take() |
|||
|
|||
|
|||
// 直接播放整个数据块,让AudioTrack自己处理时序 |
|||
audioTrack?.write(data, 0, data.size) |
|||
|
|||
} catch (e: InterruptedException) { |
|||
Log.d(TAG, "播放循环被中断") |
|||
break |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "播放循环异常: ${e.message}", e) |
|||
} |
|||
} |
|||
Log.d(TAG, "播放循环结束") |
|||
stopPlaybackInternal() |
|||
} |
|||
|
|||
/// 内部停止播放方法 |
|||
private fun stopPlaybackInternal() { |
|||
if (!isPlaying) return |
|||
isPlaying = false |
|||
|
|||
playbackQueue.clear() |
|||
|
|||
// 使用try-catch保护,因为AudioTrack状态可能不稳定 |
|||
try { |
|||
if (audioTrack?.playState == AudioTrack.PLAYSTATE_PLAYING) { |
|||
audioTrack?.pause() |
|||
audioTrack?.flush() |
|||
} |
|||
// 不要在这里调用stop,stop是用来释放资源的,pause/flush更适合流式播放的暂停 |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "暂停和清空AudioTrack失败: ${e.message}", e) |
|||
} |
|||
|
|||
onStatusChanged?.invoke(VoiceStatus.IDLE) |
|||
Log.i(TAG, "播放已通过内部调用停止") |
|||
} |
|||
|
|||
/// 停止播放(外部调用) |
|||
fun stopPlaying(): Boolean { |
|||
if (!isPlaying) { |
|||
return true |
|||
} |
|||
try { |
|||
isPlaying = false |
|||
playJob?.cancel() // 取消协程 |
|||
playJob = null |
|||
stopPlaybackInternal() |
|||
|
|||
// 恢复音频模式 |
|||
restoreAudioMode() |
|||
|
|||
return true |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "停止播放失败: ${e.message}", e) |
|||
return false |
|||
} |
|||
} |
|||
|
|||
/// 更新配置 |
|||
fun updateConfig(config: AudioConfig): Boolean { |
|||
// 如果正在录音或播放,先停止 |
|||
if (isRecording) { |
|||
stopRecording() |
|||
} |
|||
if (isPlaying) { |
|||
stopPlaying() |
|||
} |
|||
|
|||
// 释放旧的音频对象 |
|||
releaseAudioObjects() |
|||
|
|||
return initialize(config) |
|||
} |
|||
|
|||
/// 释放音频对象 |
|||
private fun releaseAudioObjects() { |
|||
try { |
|||
audioRecord?.release() |
|||
audioRecord = null |
|||
|
|||
audioTrack?.release() |
|||
audioTrack = null |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "释放音频对象失败: ${e.message}", e) |
|||
} |
|||
} |
|||
|
|||
/// 释放资源 |
|||
fun dispose() { |
|||
stopRecording() |
|||
stopPlaying() |
|||
|
|||
// 清理回音消除和恢复音频模式 |
|||
cleanupEchoCancellation() |
|||
restoreAudioMode() |
|||
|
|||
scope.cancel() |
|||
releaseAudioObjects() |
|||
playbackQueue.clear() |
|||
|
|||
Log.i(TAG, "音频管理器已释放") |
|||
} |
|||
|
|||
|
|||
|
|||
|
|||
/// 设置回音消除(总是启用) |
|||
private fun setupEchoCancellation() { |
|||
val record = audioRecord ?: return |
|||
|
|||
try { |
|||
// 设置音频模式为通信模式 |
|||
audioManager?.mode = AudioManager.MODE_IN_COMMUNICATION |
|||
|
|||
// 设置声学回音消除 |
|||
if (AcousticEchoCanceler.isAvailable()) { |
|||
acousticEchoCanceler = AcousticEchoCanceler.create(record.audioSessionId) |
|||
acousticEchoCanceler?.enabled = true |
|||
Log.i(TAG, "声学回音消除已启用") |
|||
} else { |
|||
Log.w(TAG, "设备不支持声学回音消除") |
|||
} |
|||
|
|||
// 设置噪声抑制 |
|||
if (NoiseSuppressor.isAvailable()) { |
|||
noiseSuppressor = NoiseSuppressor.create(record.audioSessionId) |
|||
noiseSuppressor?.enabled = true |
|||
Log.i(TAG, "噪声抑制已启用") |
|||
} |
|||
|
|||
// 设置自动增益控制 |
|||
if (AutomaticGainControl.isAvailable()) { |
|||
automaticGainControl = AutomaticGainControl.create(record.audioSessionId) |
|||
automaticGainControl?.enabled = true |
|||
Log.i(TAG, "自动增益控制已启用") |
|||
} |
|||
|
|||
Log.i(TAG, "回音消除和音频增强功能已全部启用") |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "设置回音消除失败: ${e.message}", e) |
|||
} |
|||
} |
|||
|
|||
/// 清理回音消除 |
|||
private fun cleanupEchoCancellation() { |
|||
try { |
|||
acousticEchoCanceler?.let { |
|||
it.enabled = false |
|||
it.release() |
|||
acousticEchoCanceler = null |
|||
Log.d(TAG, "声学回音消除已释放") |
|||
} |
|||
|
|||
noiseSuppressor?.let { |
|||
it.enabled = false |
|||
it.release() |
|||
noiseSuppressor = null |
|||
Log.d(TAG, "噪声抑制已释放") |
|||
} |
|||
|
|||
automaticGainControl?.let { |
|||
it.enabled = false |
|||
it.release() |
|||
automaticGainControl = null |
|||
Log.d(TAG, "自动增益控制已释放") |
|||
} |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "清理回音消除失败: ${e.message}", e) |
|||
} |
|||
} |
|||
|
|||
/// 恢复音频模式 |
|||
private fun restoreAudioMode() { |
|||
try { |
|||
audioManager?.mode = originalAudioMode |
|||
Log.d(TAG, "音频模式已恢复为: $originalAudioMode") |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "恢复音频模式失败: ${e.message}", e) |
|||
} |
|||
} |
|||
|
|||
fun destroy() { |
|||
Log.d(TAG, "销毁AudioManager") |
|||
stopRecording() |
|||
stopPlaying() |
|||
scope.cancel() // 确保所有协程都已取消 |
|||
|
|||
audioRecord?.release() |
|||
audioTrack?.release() |
|||
releaseAudioObjects() |
|||
playbackQueue.clear() |
|||
|
|||
Log.i(TAG, "音频管理器已释放") |
|||
} |
|||
|
|||
/// 计算音频数据的RMS值 |
|||
private fun calculateRMS(audioData: ByteArray): Float { |
|||
if (audioData.isEmpty()) return 0f |
|||
|
|||
var sum = 0.0 |
|||
// 处理16位PCM数据 |
|||
for (i in 0 until audioData.size step 2) { |
|||
if (i + 1 < audioData.size) { |
|||
// Little-endian: 低字节在前 |
|||
val sample = (audioData[i].toInt() and 0xFF) or |
|||
(audioData[i + 1].toInt() shl 8) |
|||
// 转换为有符号16位整数 |
|||
val signedSample = sample.toShort().toFloat() |
|||
// 归一化到[-1, 1] |
|||
val normalizedSample = signedSample / 32768f |
|||
sum += normalizedSample * normalizedSample |
|||
} |
|||
} |
|||
|
|||
val mean = sum / (audioData.size / 2) |
|||
return sqrt(mean).toFloat() |
|||
} |
|||
} |
|||
@ -0,0 +1,455 @@ |
|||
package com.yunqiinnovation.realtime |
|||
|
|||
import android.content.Context |
|||
import android.os.Handler |
|||
import android.os.Looper |
|||
import android.util.Log |
|||
import io.flutter.embedding.engine.plugins.FlutterPlugin |
|||
import io.flutter.plugin.common.EventChannel |
|||
import io.flutter.plugin.common.MethodCall |
|||
import io.flutter.plugin.common.MethodChannel |
|||
import io.flutter.plugin.common.MethodChannel.MethodCallHandler |
|||
import io.flutter.plugin.common.MethodChannel.Result |
|||
import kotlinx.coroutines.* |
|||
import org.json.JSONObject |
|||
|
|||
/** |
|||
* 实时语音聊天插件 |
|||
* |
|||
* 通过WebSocket连接Vocode服务器实现实时语音交互 |
|||
* 包含录音、播放、WebSocket通信等功能 |
|||
*/ |
|||
class RealtimePlugin: FlutterPlugin, MethodCallHandler, EventChannel.StreamHandler { |
|||
companion object { |
|||
private const val TAG = "RealtimePlugin" |
|||
private const val METHOD_CHANNEL = "realtime/methods" |
|||
private const val EVENT_CHANNEL = "realtime/events" |
|||
} |
|||
|
|||
private lateinit var context: Context |
|||
private lateinit var methodChannel: MethodChannel |
|||
private lateinit var eventChannel: EventChannel |
|||
private var eventSink: EventChannel.EventSink? = null |
|||
|
|||
// 核心组件:设为可空,因为它们的生命周期与页面会话绑定 |
|||
private var audioManager: RealtimeAudioManager? = null |
|||
private var webSocketManager: RealtimeWebSocketManager? = null |
|||
|
|||
// 协程作用域:需要可变,因为在重用插件时需要重新创建 |
|||
private var scope = CoroutineScope(Dispatchers.Default + SupervisorJob()) |
|||
|
|||
// 主线程Handler,用于向Flutter发送事件与回调 |
|||
private val mainHandler = Handler(Looper.getMainLooper()) |
|||
|
|||
// 配置参数 |
|||
private var serverUrl: String = "" |
|||
private var sampleRate: Int = 16000 |
|||
private var channels: Int = 1 |
|||
private var bitsPerSample: Int = 16 |
|||
|
|||
// 状态 |
|||
@Volatile |
|||
private var connectionStatus: ConnectionStatus = ConnectionStatus.DISCONNECTED |
|||
@Volatile |
|||
private var voiceStatus: VoiceStatus = VoiceStatus.IDLE |
|||
|
|||
// 清理状态标志,确保清理逻辑只执行一次 |
|||
@Volatile |
|||
private var isCleanedUp = false |
|||
private val cleanupLock = Any() |
|||
|
|||
override fun onAttachedToEngine(flutterPluginBinding: FlutterPlugin.FlutterPluginBinding) { |
|||
context = flutterPluginBinding.applicationContext |
|||
|
|||
methodChannel = MethodChannel(flutterPluginBinding.binaryMessenger, METHOD_CHANNEL) |
|||
methodChannel.setMethodCallHandler(this) |
|||
|
|||
eventChannel = EventChannel(flutterPluginBinding.binaryMessenger, EVENT_CHANNEL) |
|||
eventChannel.setStreamHandler(this) |
|||
|
|||
Log.i(TAG, "Realtime插件已附加到引擎") |
|||
} |
|||
|
|||
override fun onDetachedFromEngine(binding: FlutterPlugin.FlutterPluginBinding) { |
|||
Log.i(TAG, "插件正在从引擎分离,执行清理...") |
|||
performCleanup() |
|||
methodChannel.setMethodCallHandler(null) |
|||
eventChannel.setStreamHandler(null) |
|||
Log.i(TAG, "Realtime插件已从引擎分离") |
|||
} |
|||
|
|||
/// 设置各组件的回调 |
|||
private fun setupCallbacks() { |
|||
Log.d(TAG, "setupCallbacks: 正在为新的管理器实例设置回调...") |
|||
|
|||
// 音频管理器回调 |
|||
audioManager?.onAudioData = { data -> |
|||
// 发送到WebSocket |
|||
webSocketManager?.sendAudioData(data) |
|||
// 发送录音PCM数据到Flutter用于可视化 |
|||
val pcm16Data = convertByteArrayToPCM16(data) |
|||
sendEvent("recordingPcmData", pcm16Data) |
|||
} |
|||
|
|||
audioManager?.onStatusChanged = { status -> |
|||
voiceStatus = status |
|||
sendEvent("voiceStatusChanged", status.value) |
|||
} |
|||
|
|||
// 新增RMS回调 |
|||
audioManager?.onRmsChanged = { rms -> |
|||
sendEvent("audioRms", rms) |
|||
} |
|||
|
|||
// WebSocket管理器回调 |
|||
webSocketManager?.onConnectionStatusChanged = { status -> |
|||
connectionStatus = status |
|||
sendEvent("connectionStatusChanged", status.value) |
|||
} |
|||
|
|||
webSocketManager?.onAudioReceived = { data -> |
|||
// 添加音频数据前50字节的日志输出 |
|||
val dataPreview = if (data.size > 50) { |
|||
data.sliceArray(0..49).joinToString(" ") { "%02x".format(it) } + "..." |
|||
} else { |
|||
data.joinToString(" ") { "%02x".format(it) } |
|||
} |
|||
Log.d(TAG, "收到音频数据前50字节: $dataPreview") |
|||
Log.d(TAG, "音频数据总长度: ${data.size} 字节") |
|||
|
|||
audioManager?.playAudioData(data) |
|||
// 发送播放PCM数据到Flutter用于可视化 |
|||
val pcm16Data = convertByteArrayToPCM16(data) |
|||
sendEvent("playbackPcmData", pcm16Data) |
|||
} |
|||
|
|||
webSocketManager?.onTextReceived = { text -> |
|||
// 直接发送原始JSON文本到Flutter层处理 |
|||
sendEvent("textReceived", text) |
|||
} |
|||
|
|||
webSocketManager?.onError = { error -> |
|||
sendEvent("error", error) |
|||
} |
|||
} |
|||
|
|||
/// 发送事件到Flutter端 |
|||
private fun sendEvent(type: String, data: Any?) { |
|||
val sink = eventSink |
|||
if (sink == null) { |
|||
Log.d(TAG, "事件流已关闭,忽略事件: $type") |
|||
return |
|||
} |
|||
|
|||
val event = mapOf( |
|||
"type" to type, |
|||
"data" to data |
|||
) |
|||
|
|||
try { |
|||
mainHandler.post { |
|||
try { |
|||
sink.success(event) |
|||
} catch (e: Exception) { |
|||
Log.w(TAG, "发送事件失败: \\${e.message}") |
|||
} |
|||
} |
|||
} catch (e: Exception) { |
|||
Log.w(TAG, "发送事件时出现异常: \\${e.message}") |
|||
} |
|||
} |
|||
|
|||
override fun onMethodCall(call: MethodCall, result: Result) { |
|||
when (call.method) { |
|||
"initialize" -> handleInitialize(call, result) |
|||
"connect" -> handleConnect(result) |
|||
"disconnect" -> handleDisconnect(result) |
|||
"startRecording" -> handleStartRecording(result) |
|||
"stopRecording" -> handleStopRecording(result) |
|||
"stopPlaying" -> handleStopPlaying(result) |
|||
"getConnectionStatus" -> result.success(connectionStatus.value) |
|||
"getVoiceStatus" -> result.success(voiceStatus.value) |
|||
"sendTextMessage" -> handleSendTextMessage(call, result) |
|||
"setAudioConfig" -> handleSetAudioConfig(call, result) |
|||
"dispose" -> handleDispose(result) |
|||
else -> result.notImplemented() |
|||
} |
|||
} |
|||
|
|||
private fun handleInitialize(call: MethodCall, result: Result) { |
|||
try { |
|||
Log.d(TAG, "handleInitialize: 开始新一轮的初始化...") |
|||
|
|||
// 1. 如果之前的协程作用域已被取消,则创建一个全新的 |
|||
if (!scope.isActive) { |
|||
scope = CoroutineScope(Dispatchers.Default + SupervisorJob()) |
|||
Log.i(TAG, "handleInitialize: 检测到协程作用域已失效,已创建新实例。") |
|||
} |
|||
|
|||
// 2. 重置清理状态标志,允许下一次的清理操作 |
|||
synchronized(cleanupLock) { |
|||
if (isCleanedUp) { |
|||
isCleanedUp = false |
|||
Log.i(TAG, "handleInitialize: 清理状态已重置,插件可再次使用。") |
|||
} |
|||
} |
|||
|
|||
// 3. 为新会话创建全新的管理器实例 |
|||
Log.d(TAG, "handleInitialize: 正在创建新的AudioManager和WebSocketManager实例...") |
|||
audioManager = RealtimeAudioManager(context) |
|||
webSocketManager = RealtimeWebSocketManager() |
|||
|
|||
// 4. 为新实例设置回调 |
|||
setupCallbacks() |
|||
|
|||
val args = call.arguments as? Map<String, Any> |
|||
val serverUrl = args?.get("serverUrl") as? String |
|||
?: return result.error("INVALID_ARGUMENTS", "服务器地址不能为空", null) |
|||
|
|||
this.serverUrl = serverUrl |
|||
|
|||
args["sampleRate"]?.let { this.sampleRate = (it as Number).toInt() } |
|||
args["channels"]?.let { this.channels = (it as Number).toInt() } |
|||
args["bitsPerSample"]?.let { this.bitsPerSample = (it as Number).toInt() } |
|||
|
|||
// 初始化音频管理器 |
|||
val audioConfig = AudioConfig(sampleRate, channels, bitsPerSample) |
|||
val success = audioManager?.initialize(audioConfig) ?: false |
|||
|
|||
if (success) { |
|||
webSocketManager?.initialize(serverUrl) |
|||
Log.i(TAG, "实时语音插件初始化成功") |
|||
} else { |
|||
Log.e(TAG, "实时语音插件初始化失败") |
|||
} |
|||
|
|||
result.success(success) |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "初始化失败: ${e.message}", e) |
|||
result.error("INITIALIZATION_ERROR", "初始化失败: ${e.message}", null) |
|||
} |
|||
} |
|||
|
|||
private fun handleConnect(result: Result) { |
|||
Log.d(TAG, "handleConnect: 收到连接请求。scope是否活跃? ${scope.isActive}") |
|||
scope.launch { |
|||
try { |
|||
val success = webSocketManager?.connect() ?: false |
|||
mainHandler.post { result.success(success) } |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "连接失败: ${e.message}", e) |
|||
mainHandler.post { result.error("CONNECTION_ERROR", "连接失败: ${e.message}", null) } |
|||
} |
|||
} |
|||
} |
|||
|
|||
private fun handleDisconnect(result: Result) { |
|||
scope.launch { |
|||
try { |
|||
webSocketManager?.disconnect() |
|||
mainHandler.post { result.success(true) } |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "断开连接失败: ${e.message}", e) |
|||
mainHandler.post { result.error("DISCONNECTION_ERROR", "断开连接失败: ${e.message}", null) } |
|||
} |
|||
} |
|||
} |
|||
|
|||
private fun handleStartRecording(result: Result) { |
|||
scope.launch { |
|||
try { |
|||
val success = audioManager?.startRecording() ?: false |
|||
mainHandler.post { result.success(success) } |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "开始录音失败: ${e.message}", e) |
|||
mainHandler.post { result.error("RECORDING_ERROR", "开始录音失败: ${e.message}", null) } |
|||
} |
|||
} |
|||
} |
|||
|
|||
private fun handleStopRecording(result: Result) { |
|||
scope.launch { |
|||
try { |
|||
val success = audioManager?.stopRecording() ?: false |
|||
mainHandler.post { result.success(success) } |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "停止录音失败: ${e.message}", e) |
|||
mainHandler.post { result.error("RECORDING_ERROR", "停止录音失败: ${e.message}", null) } |
|||
} |
|||
} |
|||
} |
|||
|
|||
private fun handleStopPlaying(result: Result) { |
|||
scope.launch { |
|||
try { |
|||
val success = audioManager?.stopPlaying() ?: false |
|||
mainHandler.post { result.success(success) } |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "停止播放失败: ${e.message}", e) |
|||
mainHandler.post { result.error("PLAYBACK_ERROR", "停止播放失败: ${e.message}", null) } |
|||
} |
|||
} |
|||
} |
|||
|
|||
private fun handleSendTextMessage(call: MethodCall, result: Result) { |
|||
try { |
|||
val args = call.arguments as? Map<String, Any> |
|||
val message = args?.get("message") as? String |
|||
?: return result.error("INVALID_ARGUMENTS", "消息内容不能为空", null) |
|||
|
|||
val success = webSocketManager?.sendTextMessage(message) ?: false |
|||
result.success(success) |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "发送文本消息失败: ${e.message}", e) |
|||
result.error("MESSAGE_ERROR", "发送文本消息失败: ${e.message}", null) |
|||
} |
|||
} |
|||
|
|||
private fun handleSetAudioConfig(call: MethodCall, result: Result) { |
|||
try { |
|||
val args = call.arguments as? Map<String, Any> |
|||
?: return result.error("INVALID_ARGUMENTS", "参数无效", null) |
|||
|
|||
Log.d(TAG, "收到的音频配置参数: $args") |
|||
|
|||
args["sampleRate"]?.let { |
|||
Log.d(TAG, "sampleRate: $it, type: ${it.javaClass.name}") |
|||
this.sampleRate = (it as Number).toInt() |
|||
} |
|||
args["channels"]?.let { |
|||
Log.d(TAG, "channels: $it, type: ${it.javaClass.name}") |
|||
this.channels = (it as Number).toInt() |
|||
} |
|||
args["bitsPerSample"]?.let { |
|||
Log.d(TAG, "bitsPerSample: $it, type: ${it.javaClass.name}") |
|||
this.bitsPerSample = (it as Number).toInt() |
|||
} |
|||
|
|||
val audioConfig = AudioConfig(sampleRate, channels, bitsPerSample) |
|||
val success = audioManager?.updateConfig(audioConfig) ?: false |
|||
result.success(success) |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "设置音频配置失败: ${e.message}", e) |
|||
result.error("CONFIG_ERROR", "设置音频配置失败: ${e.message}", null) |
|||
} |
|||
} |
|||
|
|||
private fun handleDispose(result: Result) { |
|||
Log.i(TAG, "Flutter端请求释放资源...") |
|||
performCleanup() |
|||
// 立即返回,不等待异步清理完成 |
|||
result.success(null) |
|||
} |
|||
|
|||
/** |
|||
* 将字节数组转换为PCM16整数数组 |
|||
* PCM16格式:16-bit signed integers, little-endian |
|||
*/ |
|||
private fun convertByteArrayToPCM16(data: ByteArray): List<Int> { |
|||
val pcm16List = mutableListOf<Int>() |
|||
|
|||
// 确保数据长度是偶数(每2个字节一个采样) |
|||
val validSize = data.size and 0xFFFFFFFE.toInt() // 清除最低位,确保偶数 |
|||
|
|||
// 每2个字节组成一个16-bit采样 |
|||
for (i in 0 until validSize step 2) { |
|||
if (i + 1 < data.size) { // 安全检查 |
|||
// Little-endian: 低字节在前 |
|||
val low = data[i].toInt() and 0xFF |
|||
val high = data[i + 1].toInt() shl 8 |
|||
val sample = (high or low).toShort().toInt() // 转换为有符号16位整数 |
|||
pcm16List.add(sample) |
|||
} |
|||
} |
|||
|
|||
return pcm16List |
|||
} |
|||
|
|||
/** |
|||
* 执行核心清理操作。 |
|||
* 此方法是幂等的(可重复调用)和线程安全的。 |
|||
*/ |
|||
private fun performCleanup() { |
|||
synchronized(cleanupLock) { |
|||
if (isCleanedUp) { |
|||
Log.d(TAG, "资源已清理,跳过。") |
|||
return |
|||
} |
|||
Log.i(TAG, "开始执行核心资源清理...") |
|||
|
|||
// 立即标记,防止重入 |
|||
isCleanedUp = true |
|||
|
|||
// 立即停止接收事件 |
|||
eventSink = null |
|||
|
|||
// 在后台IO线程中执行所有耗时操作,避免阻塞主线程 |
|||
CoroutineScope(Dispatchers.IO).launch { |
|||
runCatching { audioManager?.stopRecording() } |
|||
.onFailure { Log.w(TAG, "停止录音时异常: ${it.message}") } |
|||
|
|||
runCatching { audioManager?.stopPlaying() } |
|||
.onFailure { Log.w(TAG, "停止播放时异常: ${it.message}") } |
|||
|
|||
runCatching { webSocketManager?.disconnect() } |
|||
.onFailure { Log.w(TAG, "断开WebSocket时异常: ${it.message}") } |
|||
|
|||
// 短暂延迟,以确保挂起的操作有时间完成 |
|||
// delay(100L) |
|||
|
|||
runCatching { audioManager?.dispose() } |
|||
.onFailure { Log.w(TAG, "释放音频管理器时异常: ${it.message}") } |
|||
|
|||
runCatching { webSocketManager?.dispose() } |
|||
.onFailure { Log.w(TAG, "释放WebSocket管理器时异常: ${it.message}") } |
|||
|
|||
runCatching { |
|||
scope.cancel() |
|||
Log.i(TAG, "插件协程作用域已取消。") |
|||
}.onFailure { Log.w(TAG, "取消协程作用域时异常: ${it.message}") } |
|||
|
|||
// 清理完成,将实例置空 |
|||
audioManager = null |
|||
webSocketManager = null |
|||
|
|||
Log.i(TAG, "核心资源清理完成。") |
|||
} |
|||
} |
|||
} |
|||
|
|||
// EventChannel.StreamHandler 实现 |
|||
override fun onListen(arguments: Any?, events: EventChannel.EventSink?) { |
|||
eventSink = events |
|||
Log.i(TAG, "事件流监听已开始") |
|||
} |
|||
|
|||
override fun onCancel(arguments: Any?) { |
|||
eventSink = null |
|||
Log.i(TAG, "事件流监听已取消") |
|||
} |
|||
} |
|||
|
|||
// 枚举定义 |
|||
enum class ConnectionStatus(val value: String) { |
|||
DISCONNECTED("disconnected"), |
|||
CONNECTING("connecting"), |
|||
CONNECTED("connected"), |
|||
ERROR("error") |
|||
} |
|||
|
|||
enum class VoiceStatus(val value: String) { |
|||
IDLE("idle"), |
|||
RECORDING("recording"), |
|||
PROCESSING("processing"), |
|||
PLAYING("playing") |
|||
} |
|||
|
|||
// 音频配置 |
|||
data class AudioConfig( |
|||
val sampleRate: Int, |
|||
val channels: Int, |
|||
val bitsPerSample: Int |
|||
) { |
|||
val frameSize: Int |
|||
get() = (sampleRate * channels * bitsPerSample / 8 * 20) / 1000 // 20ms帧 |
|||
} |
|||
@ -0,0 +1,303 @@ |
|||
package com.yunqiinnovation.realtime |
|||
|
|||
import android.util.Log |
|||
import kotlinx.coroutines.* |
|||
import okhttp3.* |
|||
import okio.ByteString |
|||
import java.util.concurrent.TimeUnit |
|||
|
|||
/** |
|||
* 实时WebSocket管理器 |
|||
* |
|||
* 负责与Vocode服务器的WebSocket通信,对应Android示例中的OkHttp WebSocket功能 |
|||
*/ |
|||
class RealtimeWebSocketManager : WebSocketListener() { |
|||
companion object { |
|||
private const val TAG = "RealtimeWebSocketManager" |
|||
private const val CONNECT_TIMEOUT = 30L |
|||
private const val READ_TIMEOUT = 60L |
|||
private const val WRITE_TIMEOUT = 30L |
|||
} |
|||
|
|||
// OkHttp客户端和WebSocket |
|||
private var okHttpClient: OkHttpClient? = null |
|||
private var webSocket: WebSocket? = null |
|||
|
|||
// 服务器URL |
|||
private var serverUrl: String = "" |
|||
|
|||
// 连接状态 |
|||
@Volatile |
|||
private var isConnected = false |
|||
@Volatile |
|||
private var isConnecting = false |
|||
|
|||
// 回调 |
|||
var onConnectionStatusChanged: ((ConnectionStatus) -> Unit)? = null |
|||
var onAudioReceived: ((ByteArray) -> Unit)? = null |
|||
var onTextReceived: ((String) -> Unit)? = null |
|||
var onError: ((String) -> Unit)? = null |
|||
|
|||
// 协程作用域 |
|||
private val scope = CoroutineScope(Dispatchers.IO + SupervisorJob()) |
|||
|
|||
/// 初始化WebSocket管理器 |
|||
fun initialize(serverUrl: String) { |
|||
Log.d(TAG, "initialize: 开始初始化。当前httpClient是否为null? ${okHttpClient == null}") |
|||
this.serverUrl = serverUrl |
|||
|
|||
// 创建OkHttp客户端 |
|||
okHttpClient = OkHttpClient.Builder() |
|||
.connectTimeout(CONNECT_TIMEOUT, TimeUnit.SECONDS) |
|||
.readTimeout(READ_TIMEOUT, TimeUnit.SECONDS) |
|||
.writeTimeout(WRITE_TIMEOUT, TimeUnit.SECONDS) |
|||
.retryOnConnectionFailure(true) |
|||
.build() |
|||
|
|||
Log.i(TAG, "initialize: WebSocket管理器初始化完成。服务器地址: $serverUrl") |
|||
} |
|||
|
|||
/// 连接到服务器 |
|||
suspend fun connect(): Boolean = withContext(Dispatchers.IO) { |
|||
Log.d(TAG, "connect: 尝试连接。当前状态 - isConnected: $isConnected, isConnecting: $isConnecting") |
|||
if (isConnected || isConnecting) { |
|||
Log.w(TAG, "connect: 连接请求被拒绝,因为已有连接或正在连接中。") |
|||
return@withContext false |
|||
} |
|||
|
|||
val client = okHttpClient ?: run { |
|||
Log.e(TAG, "connect: 连接失败,因为OkHttp客户端为null。请确保已调用initialize()。") |
|||
onError?.invoke("客户端未初始化") |
|||
return@withContext false |
|||
} |
|||
|
|||
Log.d(TAG, "connect: OkHttpClient实例存在,准备发起新连接。") |
|||
|
|||
try { |
|||
isConnecting = true |
|||
onConnectionStatusChanged?.invoke(ConnectionStatus.CONNECTING) |
|||
|
|||
val request = Request.Builder() |
|||
.url(serverUrl) |
|||
.build() |
|||
|
|||
Log.d(TAG, "connect: 正在创建新的WebSocket实例...") |
|||
webSocket = client.newWebSocket(request, this@RealtimeWebSocketManager) |
|||
|
|||
// 等待连接建立 |
|||
var waitTime = 0 |
|||
while (isConnecting && waitTime < CONNECT_TIMEOUT * 1000) { |
|||
delay(100) |
|||
waitTime += 100 |
|||
} |
|||
|
|||
if (isConnected) { |
|||
Log.i(TAG, "connect: WebSocket连接成功") |
|||
return@withContext true |
|||
} else { |
|||
Log.e(TAG, "connect: WebSocket连接超时") |
|||
isConnecting = false |
|||
onConnectionStatusChanged?.invoke(ConnectionStatus.ERROR) |
|||
return@withContext false |
|||
} |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "WebSocket连接失败: ${e.message}", e) |
|||
isConnecting = false |
|||
onConnectionStatusChanged?.invoke(ConnectionStatus.ERROR) |
|||
onError?.invoke("连接失败: ${e.message}") |
|||
return@withContext false |
|||
} |
|||
} |
|||
|
|||
/// 断开连接 |
|||
fun disconnect() { |
|||
try { |
|||
isConnected = false |
|||
isConnecting = false |
|||
|
|||
webSocket?.close(1000, "正常关闭") |
|||
webSocket = null |
|||
|
|||
onConnectionStatusChanged?.invoke(ConnectionStatus.DISCONNECTED) |
|||
Log.i(TAG, "WebSocket已断开连接") |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "断开WebSocket连接失败: ${e.message}", e) |
|||
} |
|||
} |
|||
|
|||
/// 发送音频数据 |
|||
fun sendAudioData(data: ByteArray) { |
|||
if (!isConnected) { |
|||
return |
|||
} |
|||
|
|||
try { |
|||
// 将音频数据编码为Base64并包装成Vocode WebSocket格式 |
|||
val base64AudioData = android.util.Base64.encodeToString(data, android.util.Base64.NO_WRAP) |
|||
val audioMessage = org.json.JSONObject().apply { |
|||
put("type", "websocket_audio") |
|||
put("data", base64AudioData) |
|||
} |
|||
|
|||
val success = webSocket?.send(audioMessage.toString()) ?: false |
|||
|
|||
if (!success) { |
|||
Log.w(TAG, "发送音频数据失败") |
|||
} |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "发送音频数据异常: ${e.message}", e) |
|||
} |
|||
} |
|||
|
|||
/// 发送文本消息 |
|||
fun sendTextMessage(message: String): Boolean { |
|||
if (!isConnected) { |
|||
Log.w(TAG, "WebSocket未连接,无法发送文本消息") |
|||
return false |
|||
} |
|||
|
|||
return try { |
|||
val success = webSocket?.send(message) ?: false |
|||
if (!success) { |
|||
Log.w(TAG, "发送文本消息失败") |
|||
} |
|||
success |
|||
} catch (e: Exception) { |
|||
Log.e(TAG, "发送文本消息异常: ${e.message}", e) |
|||
false |
|||
} |
|||
} |
|||
|
|||
/// 释放资源 |
|||
fun dispose() { |
|||
disconnect() |
|||
|
|||
// 立即取消scope,不等待 |
|||
try { |
|||
scope.cancel() |
|||
} catch (e: Exception) { |
|||
Log.w(TAG, "取消scope时出现异常: ${e.message}") |
|||
} |
|||
|
|||
// 立即强制关闭WebSocket |
|||
try { |
|||
webSocket?.close(1001, "强制关闭") |
|||
webSocket = null |
|||
} catch (e: Exception) { |
|||
Log.w(TAG, "强制关闭WebSocket时出现异常: ${e.message}") |
|||
} |
|||
|
|||
// 异步关闭线程池,设置更短的超时时间 |
|||
okHttpClient?.dispatcher?.executorService?.let { executor -> |
|||
Thread { |
|||
try { |
|||
executor.shutdownNow() // 立即强制关闭 |
|||
// 最多等待1秒,避免长时间等待 |
|||
if (!executor.awaitTermination(1, TimeUnit.SECONDS)) { |
|||
Log.w(TAG, "线程池强制关闭超时,放弃等待") |
|||
} |
|||
} catch (e: Exception) { |
|||
Log.w(TAG, "关闭线程池时出现异常: ${e.message}") |
|||
} |
|||
}.start() |
|||
} |
|||
okHttpClient = null |
|||
|
|||
Log.i(TAG, "WebSocket管理器已释放") |
|||
} |
|||
|
|||
// WebSocketListener 回调实现 |
|||
override fun onOpen(webSocket: WebSocket, response: Response) { |
|||
Log.i(TAG, "WebSocket连接已建立") |
|||
isConnected = true |
|||
isConnecting = false |
|||
onConnectionStatusChanged?.invoke(ConnectionStatus.CONNECTED) |
|||
} |
|||
|
|||
override fun onMessage(webSocket: WebSocket, text: String) { |
|||
// 添加收到文本消息的日志输出,显示消息类型和前20字符 |
|||
val textPreview = if (text.length > 50) text.substring(0, 50) + "..." else text |
|||
Log.d(TAG, "收到文本消息前20字符: $textPreview") |
|||
|
|||
try { |
|||
// 尝试解析为JSON来检查消息类型 |
|||
val json = org.json.JSONObject(text) |
|||
val messageType = json.optString("type", "") |
|||
|
|||
// 添加消息类型的日志输出 |
|||
Log.d(TAG, "消息类型: $messageType") |
|||
|
|||
when (messageType) { |
|||
"websocket_audio" -> { |
|||
// 处理音频消息 |
|||
val audioDataString = json.optString("data", "") |
|||
if (audioDataString.isNotEmpty()) { |
|||
// 将Base64编码的音频数据解码为字节数组 |
|||
val audioData = android.util.Base64.decode(audioDataString, android.util.Base64.DEFAULT) |
|||
onAudioReceived?.invoke(audioData) |
|||
} |
|||
} |
|||
"websocket_transcript" -> { |
|||
// 处理统一的文本消息格式 |
|||
// 包含sender、is_final、timestamp等字段 |
|||
onTextReceived?.invoke(text) |
|||
} |
|||
"websocket_ready" -> { |
|||
// 连接就绪消息 |
|||
Log.i(TAG, "WebSocket连接就绪") |
|||
} |
|||
"websocket_stop" -> { |
|||
// 会话停止消息 |
|||
Log.i(TAG, "会话已结束") |
|||
} |
|||
"websocket_start" -> { |
|||
// 会话开始消息 |
|||
Log.i(TAG, "会话已开始") |
|||
} |
|||
else -> { |
|||
// 其他消息类型,当作普通文本消息处理 |
|||
onTextReceived?.invoke(text) |
|||
} |
|||
} |
|||
} catch (e: Exception) { |
|||
// 如果不是JSON格式,当作普通文本消息处理 |
|||
Log.d(TAG, "非JSON格式消息,当作普通文本处理") |
|||
onTextReceived?.invoke(text) |
|||
} |
|||
} |
|||
|
|||
override fun onMessage(webSocket: WebSocket, bytes: ByteString) { |
|||
val audioData = bytes.toByteArray() |
|||
onAudioReceived?.invoke(audioData) |
|||
} |
|||
|
|||
override fun onClosing(webSocket: WebSocket, code: Int, reason: String) { |
|||
Log.i(TAG, "WebSocket正在关闭,代码: $code, 原因: $reason") |
|||
webSocket.close(1000, null) |
|||
} |
|||
|
|||
override fun onClosed(webSocket: WebSocket, code: Int, reason: String) { |
|||
Log.i(TAG, "WebSocket已关闭,代码: $code, 原因: $reason") |
|||
isConnected = false |
|||
isConnecting = false |
|||
onConnectionStatusChanged?.invoke(ConnectionStatus.DISCONNECTED) |
|||
} |
|||
|
|||
override fun onFailure(webSocket: WebSocket, t: Throwable, response: Response?) { |
|||
val errorMessage = when { |
|||
t.message?.contains("Unable to parse TLS packet header") == true -> |
|||
"SSL错误:请检查服务器是否支持wss://,本地开发请使用ws://" |
|||
t.message?.contains("Connection refused") == true -> |
|||
"连接被拒绝:请检查服务器是否运行在指定端口" |
|||
t.message?.contains("timeout") == true -> |
|||
"连接超时:请检查网络连接和服务器地址" |
|||
else -> "连接失败: ${t.message}" |
|||
} |
|||
|
|||
Log.e(TAG, "WebSocket连接失败: ${t.message}", t) |
|||
isConnected = false |
|||
isConnecting = false |
|||
onConnectionStatusChanged?.invoke(ConnectionStatus.ERROR) |
|||
onError?.invoke(errorMessage) |
|||
} |
|||
} |
|||
@ -0,0 +1,274 @@ |
|||
# Realtime Plugin 使用示例 |
|||
|
|||
## 基本用法 |
|||
|
|||
```dart |
|||
import 'package:realtime/realtime.dart'; |
|||
|
|||
class VoiceCallPage extends StatefulWidget { |
|||
@override |
|||
_VoiceCallPageState createState() => _VoiceCallPageState(); |
|||
} |
|||
|
|||
class _VoiceCallPageState extends State<VoiceCallPage> { |
|||
final RealtimeService _realtimeService = RealtimeService(); |
|||
StreamSubscription<RealtimeEvent>? _eventSubscription; |
|||
|
|||
ConnectionStatus _connectionStatus = ConnectionStatus.disconnected; |
|||
VoiceStatus _voiceStatus = VoiceStatus.idle; |
|||
|
|||
@override |
|||
void initState() { |
|||
super.initState(); |
|||
_setupEventListener(); |
|||
_initializeService(); |
|||
} |
|||
|
|||
void _setupEventListener() { |
|||
_eventSubscription = _realtimeService.eventStream.listen((event) { |
|||
switch (event.type) { |
|||
case 'connectionStatusChanged': |
|||
setState(() { |
|||
_connectionStatus = _parseConnectionStatus(event.data); |
|||
}); |
|||
break; |
|||
case 'voiceStatusChanged': |
|||
setState(() { |
|||
_voiceStatus = _parseVoiceStatus(event.data); |
|||
}); |
|||
break; |
|||
case 'textReceived': |
|||
print('收到文本: ${event.data}'); |
|||
break; |
|||
case 'error': |
|||
print('错误: ${event.data}'); |
|||
break; |
|||
} |
|||
}); |
|||
} |
|||
|
|||
Future<void> _initializeService() async { |
|||
try { |
|||
final success = await _realtimeService.initialize( |
|||
serverUrl: 'wss://your-vocode-server.com/ws', |
|||
sampleRate: 16000, |
|||
channels: 1, |
|||
bitsPerSample: 16, |
|||
); |
|||
|
|||
if (success) { |
|||
print('Realtime服务初始化成功'); |
|||
} else { |
|||
print('Realtime服务初始化失败'); |
|||
} |
|||
} catch (e) { |
|||
print('初始化错误: $e'); |
|||
} |
|||
} |
|||
|
|||
Future<void> _connect() async { |
|||
try { |
|||
final success = await _realtimeService.connect(); |
|||
if (!success) { |
|||
print('连接失败'); |
|||
} |
|||
} catch (e) { |
|||
print('连接错误: $e'); |
|||
} |
|||
} |
|||
|
|||
Future<void> _disconnect() async { |
|||
try { |
|||
await _realtimeService.disconnect(); |
|||
} catch (e) { |
|||
print('断开连接错误: $e'); |
|||
} |
|||
} |
|||
|
|||
Future<void> _startRecording() async { |
|||
try { |
|||
final success = await _realtimeService.startRecording(); |
|||
if (!success) { |
|||
print('开始录音失败'); |
|||
} |
|||
} catch (e) { |
|||
print('录音错误: $e'); |
|||
} |
|||
} |
|||
|
|||
Future<void> _stopRecording() async { |
|||
try { |
|||
final success = await _realtimeService.stopRecording(); |
|||
if (!success) { |
|||
print('停止录音失败'); |
|||
} |
|||
} catch (e) { |
|||
print('停止录音错误: $e'); |
|||
} |
|||
} |
|||
|
|||
ConnectionStatus _parseConnectionStatus(String status) { |
|||
switch (status) { |
|||
case 'connected': return ConnectionStatus.connected; |
|||
case 'connecting': return ConnectionStatus.connecting; |
|||
case 'disconnected': return ConnectionStatus.disconnected; |
|||
case 'error': return ConnectionStatus.error; |
|||
default: return ConnectionStatus.disconnected; |
|||
} |
|||
} |
|||
|
|||
VoiceStatus _parseVoiceStatus(String status) { |
|||
switch (status) { |
|||
case 'recording': return VoiceStatus.recording; |
|||
case 'processing': return VoiceStatus.processing; |
|||
case 'playing': return VoiceStatus.playing; |
|||
case 'idle': return VoiceStatus.idle; |
|||
default: return VoiceStatus.idle; |
|||
} |
|||
} |
|||
|
|||
@override |
|||
Widget build(BuildContext context) { |
|||
return Scaffold( |
|||
appBar: AppBar( |
|||
title: Text('实时语音聊天'), |
|||
), |
|||
body: Center( |
|||
child: Column( |
|||
mainAxisAlignment: MainAxisAlignment.center, |
|||
children: [ |
|||
// 连接状态显示 |
|||
Text('连接状态: ${_connectionStatus.name}'), |
|||
SizedBox(height: 20), |
|||
|
|||
// 语音状态显示 |
|||
Text('语音状态: ${_voiceStatus.name}'), |
|||
SizedBox(height: 40), |
|||
|
|||
// 连接按钮 |
|||
ElevatedButton( |
|||
onPressed: _connectionStatus == ConnectionStatus.disconnected |
|||
? _connect |
|||
: _disconnect, |
|||
child: Text(_connectionStatus == ConnectionStatus.disconnected |
|||
? '连接' |
|||
: '断开'), |
|||
), |
|||
SizedBox(height: 20), |
|||
|
|||
// 录音按钮 |
|||
ElevatedButton( |
|||
onPressed: _connectionStatus == ConnectionStatus.connected |
|||
? (_voiceStatus == VoiceStatus.recording |
|||
? _stopRecording |
|||
: _startRecording) |
|||
: null, |
|||
child: Text(_voiceStatus == VoiceStatus.recording |
|||
? '停止录音' |
|||
: '开始录音'), |
|||
), |
|||
], |
|||
), |
|||
), |
|||
); |
|||
} |
|||
|
|||
@override |
|||
void dispose() { |
|||
_eventSubscription?.cancel(); |
|||
_realtimeService.dispose(); |
|||
super.dispose(); |
|||
} |
|||
} |
|||
``` |
|||
|
|||
## GetX控制器中的使用 |
|||
|
|||
```dart |
|||
import 'package:get/get.dart'; |
|||
import 'package:realtime/realtime.dart'; |
|||
|
|||
class VoiceController extends GetxController { |
|||
final RealtimeService _realtimeService = RealtimeService(); |
|||
|
|||
final RxBool isConnected = false.obs; |
|||
final RxBool isRecording = false.obs; |
|||
|
|||
StreamSubscription<RealtimeEvent>? _eventSubscription; |
|||
|
|||
@override |
|||
void onInit() { |
|||
super.onInit(); |
|||
_setupEventListener(); |
|||
_initializeService(); |
|||
} |
|||
|
|||
void _setupEventListener() { |
|||
_eventSubscription = _realtimeService.eventStream.listen((event) { |
|||
switch (event.type) { |
|||
case 'connectionStatusChanged': |
|||
isConnected.value = event.data == 'connected'; |
|||
break; |
|||
case 'voiceStatusChanged': |
|||
isRecording.value = event.data == 'recording'; |
|||
break; |
|||
} |
|||
}); |
|||
} |
|||
|
|||
Future<void> _initializeService() async { |
|||
await _realtimeService.initialize( |
|||
serverUrl: 'wss://your-server/ws', |
|||
); |
|||
} |
|||
|
|||
Future<void> toggleConnection() async { |
|||
if (isConnected.value) { |
|||
await _realtimeService.disconnect(); |
|||
} else { |
|||
await _realtimeService.connect(); |
|||
} |
|||
} |
|||
|
|||
Future<void> toggleRecording() async { |
|||
if (isRecording.value) { |
|||
await _realtimeService.stopRecording(); |
|||
} else { |
|||
await _realtimeService.startRecording(); |
|||
} |
|||
} |
|||
|
|||
@override |
|||
void onClose() { |
|||
_eventSubscription?.cancel(); |
|||
_realtimeService.dispose(); |
|||
super.onClose(); |
|||
} |
|||
} |
|||
``` |
|||
|
|||
## 权限配置 |
|||
|
|||
### iOS |
|||
在 `ios/Runner/Info.plist` 中添加: |
|||
|
|||
```xml |
|||
<key>NSMicrophoneUsageDescription</key> |
|||
<string>应用需要麦克风权限进行语音录制</string> |
|||
``` |
|||
|
|||
## 服务器配置示例 |
|||
|
|||
需要一个支持WebSocket的Vocode服务器,具体实现可参考Vocode官方文档。 |
|||
|
|||
服务器需要: |
|||
1. 接收16kHz/16-bit/单声道的PCM音频数据 |
|||
2. 返回相同格式的音频数据 |
|||
3. 支持文本消息交换 |
|||
|
|||
## 注意事项 |
|||
|
|||
1. 确保网络连接稳定 |
|||
2. 音频格式必须与服务器保持一致 |
|||
3. 及时处理事件流中的错误 |
|||
4. 在适当时机释放资源 |
|||
@ -0,0 +1,18 @@ |
|||
// swift-tools-version: 5.9 |
|||
import PackageDescription |
|||
|
|||
let package = Package( |
|||
name: "realtime", |
|||
platforms: [.iOS("18.0")], |
|||
products: [ |
|||
.library(name: "realtime", targets: ["realtime"]) |
|||
], |
|||
dependencies: [], |
|||
targets: [ |
|||
.target( |
|||
name: "realtime", |
|||
dependencies: [], |
|||
path: "Sources/realtime" |
|||
) |
|||
] |
|||
) |
|||
@ -0,0 +1,344 @@ |
|||
import AVFoundation |
|||
import os.log |
|||
import Accelerate |
|||
|
|||
/** |
|||
* 实时音频管理器 |
|||
* |
|||
* 负责音频的录制和播放,对应Android示例中的AudioRecord和AudioTrack功能 |
|||
*/ |
|||
class RealtimeAudioManager { |
|||
|
|||
private let log = OSLog(subsystem: "com.yunqiinnovation.realtime", category: "RealtimeAudioManager") |
|||
|
|||
// 音频配置 |
|||
private var audioConfig: AudioConfig? |
|||
|
|||
// 录音相关 |
|||
private var audioEngine: AVAudioEngine? |
|||
private var inputNode: AVAudioInputNode? |
|||
private var recordingFormat: AVAudioFormat? |
|||
private var isRecording = false |
|||
|
|||
// 播放相关 |
|||
private var audioPlayer: AVAudioPlayerNode? |
|||
private var playbackFormat: AVAudioFormat? |
|||
private var playbackQueue: DispatchQueue |
|||
private var audioBufferQueue: [AVAudioPCMBuffer] = [] |
|||
private var isPlaying = false |
|||
|
|||
// 回调 |
|||
var onAudioData: ((Data) -> Void)? |
|||
var onStatusChanged: ((VoiceStatus) -> Void)? |
|||
var onRmsChanged: ((Float) -> Void)? |
|||
|
|||
// 录音缓冲区 |
|||
private var recordingBuffer: AVAudioPCMBuffer? |
|||
private var frameSize: Int = 320 // 20ms @ 16kHz |
|||
|
|||
init() { |
|||
playbackQueue = DispatchQueue(label: "com.realtime.playback", qos: .userInitiated) |
|||
setupAudioSession() |
|||
} |
|||
|
|||
/// 设置音频会话 |
|||
private func setupAudioSession() { |
|||
do { |
|||
let session = AVAudioSession.sharedInstance() |
|||
try session.setCategory(.playAndRecord, |
|||
mode: .default, |
|||
options: [.defaultToSpeaker, .allowBluetooth]) |
|||
try session.setActive(true) |
|||
os_log("音频会话设置成功", log: log, type: .info) |
|||
} catch { |
|||
os_log("音频会话设置失败: %@", log: log, type: .error, error.localizedDescription) |
|||
} |
|||
} |
|||
|
|||
/// 初始化音频管理器 |
|||
func initialize(config: AudioConfig) -> Bool { |
|||
self.audioConfig = config |
|||
self.frameSize = config.frameSize |
|||
|
|||
guard setupAudioEngine(config: config) else { |
|||
os_log("音频引擎初始化失败", log: log, type: .error) |
|||
return false |
|||
} |
|||
|
|||
guard setupAudioPlayer(config: config) else { |
|||
os_log("音频播放器初始化失败", log: log, type: .error) |
|||
return false |
|||
} |
|||
|
|||
os_log("音频管理器初始化成功", log: log, type: .info) |
|||
return true |
|||
} |
|||
|
|||
/// 设置音频引擎(录音) |
|||
private func setupAudioEngine(config: AudioConfig) -> Bool { |
|||
audioEngine = AVAudioEngine() |
|||
|
|||
guard let audioEngine = audioEngine else { return false } |
|||
|
|||
inputNode = audioEngine.inputNode |
|||
|
|||
// 创建录音格式:16kHz, 16-bit, 单声道 |
|||
recordingFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, |
|||
sampleRate: Double(config.sampleRate), |
|||
channels: AVAudioChannelCount(config.channels), |
|||
interleaved: true) |
|||
|
|||
guard let recordingFormat = recordingFormat else { |
|||
os_log("无法创建录音格式", log: log, type: .error) |
|||
return false |
|||
} |
|||
|
|||
// 创建录音缓冲区 |
|||
let frameCount = AVAudioFrameCount(config.frameSize / (config.bitsPerSample / 8)) |
|||
recordingBuffer = AVAudioPCMBuffer(pcmFormat: recordingFormat, frameCapacity: frameCount) |
|||
|
|||
return true |
|||
} |
|||
|
|||
/// 设置音频播放器 |
|||
private func setupAudioPlayer(config: AudioConfig) -> Bool { |
|||
guard let audioEngine = audioEngine else { return false } |
|||
|
|||
audioPlayer = AVAudioPlayerNode() |
|||
|
|||
// 创建播放格式:16kHz, 16-bit, 单声道 |
|||
playbackFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, |
|||
sampleRate: Double(config.sampleRate), |
|||
channels: AVAudioChannelCount(config.channels), |
|||
interleaved: true) |
|||
|
|||
guard let audioPlayer = audioPlayer, |
|||
let playbackFormat = playbackFormat else { |
|||
os_log("无法创建播放格式", log: log, type: .error) |
|||
return false |
|||
} |
|||
|
|||
// 连接播放器到音频引擎 |
|||
audioEngine.attach(audioPlayer) |
|||
audioEngine.connect(audioPlayer, to: audioEngine.outputNode, format: playbackFormat) |
|||
|
|||
return true |
|||
} |
|||
|
|||
/// 开始录音 |
|||
func startRecording() -> Bool { |
|||
guard let audioEngine = audioEngine, |
|||
let inputNode = inputNode, |
|||
let recordingFormat = recordingFormat, |
|||
!isRecording else { |
|||
return false |
|||
} |
|||
|
|||
// 安装录音tap |
|||
inputNode.installTap(onBus: 0, bufferSize: AVAudioFrameCount(frameSize / 2), format: recordingFormat) { [weak self] buffer, time in |
|||
self?.processRecordedAudio(buffer: buffer) |
|||
} |
|||
|
|||
do { |
|||
try audioEngine.start() |
|||
isRecording = true |
|||
onStatusChanged?(.recording) |
|||
os_log("开始录音", log: log, type: .info) |
|||
return true |
|||
} catch { |
|||
os_log("开始录音失败: %@", log: log, type: .error, error.localizedDescription) |
|||
return false |
|||
} |
|||
} |
|||
|
|||
/// 处理录制的音频数据 |
|||
private func processRecordedAudio(buffer: AVAudioPCMBuffer) { |
|||
guard let channelData = buffer.int16ChannelData?[0], |
|||
buffer.frameLength > 0 else { |
|||
return |
|||
} |
|||
|
|||
// 确保数据长度匹配frameSize |
|||
let dataLength = min(Int(buffer.frameLength) * 2, frameSize) // 2 bytes per sample |
|||
|
|||
// 转换为Data |
|||
let data = Data(bytes: channelData, count: dataLength) |
|||
|
|||
// 计算RMS值 |
|||
let rms = calculateRMS(buffer: buffer) |
|||
onRmsChanged?(rms) |
|||
|
|||
// 回调音频数据 |
|||
onAudioData?(data) |
|||
} |
|||
|
|||
/// 停止录音 |
|||
func stopRecording() -> Bool { |
|||
guard let audioEngine = audioEngine, |
|||
let inputNode = inputNode, |
|||
isRecording else { |
|||
return false |
|||
} |
|||
|
|||
inputNode.removeTap(onBus: 0) |
|||
audioEngine.stop() |
|||
isRecording = false |
|||
onStatusChanged?(.idle) |
|||
os_log("停止录音", log: log, type: .info) |
|||
return true |
|||
} |
|||
|
|||
/// 播放音频数据 |
|||
func playAudioData(_ data: Data) { |
|||
guard let playbackFormat = playbackFormat, |
|||
let audioPlayer = audioPlayer else { |
|||
return |
|||
} |
|||
|
|||
playbackQueue.async { [weak self] in |
|||
self?.enqueueAudioData(data, format: playbackFormat, player: audioPlayer) |
|||
} |
|||
} |
|||
|
|||
/// 将音频数据加入播放队列 |
|||
private func enqueueAudioData(_ data: Data, format: AVAudioFormat, player: AVAudioPlayerNode) { |
|||
// 创建PCM缓冲区 |
|||
let frameCount = AVAudioFrameCount(data.count / 2) // 16-bit = 2 bytes per sample |
|||
|
|||
guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: frameCount) else { |
|||
os_log("无法创建播放缓冲区", log: log, type: .error) |
|||
return |
|||
} |
|||
|
|||
buffer.frameLength = frameCount |
|||
|
|||
// 复制数据到缓冲区 |
|||
guard let channelData = buffer.int16ChannelData?[0] else { |
|||
return |
|||
} |
|||
|
|||
data.withUnsafeBytes { bytes in |
|||
let int16Pointer = bytes.bindMemory(to: Int16.self) |
|||
channelData.update(from: int16Pointer.baseAddress!, count: Int(frameCount)) |
|||
} |
|||
|
|||
// 计算播放数据的RMS值 |
|||
let rms = calculateRMS(buffer: buffer) |
|||
DispatchQueue.main.async { [weak self] in |
|||
self?.onRmsChanged?(rms) |
|||
} |
|||
|
|||
// 加入播放队列 |
|||
audioBufferQueue.append(buffer) |
|||
|
|||
// 如果没在播放,开始播放 |
|||
if !isPlaying { |
|||
startPlayback() |
|||
} |
|||
} |
|||
|
|||
/// 开始播放 |
|||
private func startPlayback() { |
|||
guard let audioEngine = audioEngine, |
|||
let audioPlayer = audioPlayer, |
|||
!audioBufferQueue.isEmpty else { |
|||
return |
|||
} |
|||
|
|||
if !audioEngine.isRunning { |
|||
do { |
|||
try audioEngine.start() |
|||
} catch { |
|||
os_log("启动音频引擎失败: %@", log: log, type: .error, error.localizedDescription) |
|||
return |
|||
} |
|||
} |
|||
|
|||
if !audioPlayer.isPlaying { |
|||
audioPlayer.play() |
|||
} |
|||
|
|||
isPlaying = true |
|||
onStatusChanged?(.playing) |
|||
|
|||
// 播放队列中的音频 |
|||
playNextBuffer() |
|||
} |
|||
|
|||
/// 播放下一个缓冲区 |
|||
private func playNextBuffer() { |
|||
guard let audioPlayer = audioPlayer, |
|||
!audioBufferQueue.isEmpty else { |
|||
isPlaying = false |
|||
onStatusChanged?(.idle) |
|||
return |
|||
} |
|||
|
|||
let buffer = audioBufferQueue.removeFirst() |
|||
|
|||
audioPlayer.scheduleBuffer(buffer) { [weak self] in |
|||
DispatchQueue.main.async { |
|||
self?.playNextBuffer() |
|||
} |
|||
} |
|||
} |
|||
|
|||
/// 停止播放 |
|||
func stopPlaying() -> Bool { |
|||
guard let audioPlayer = audioPlayer else { |
|||
return false |
|||
} |
|||
|
|||
audioPlayer.stop() |
|||
audioBufferQueue.removeAll() |
|||
isPlaying = false |
|||
onStatusChanged?(.idle) |
|||
os_log("停止播放", log: log, type: .info) |
|||
return true |
|||
} |
|||
|
|||
/// 更新配置 |
|||
func updateConfig(config: AudioConfig) -> Bool { |
|||
// 如果正在录音或播放,先停止 |
|||
if isRecording { |
|||
_ = stopRecording() |
|||
} |
|||
if isPlaying { |
|||
_ = stopPlaying() |
|||
} |
|||
|
|||
return initialize(config: config) |
|||
} |
|||
|
|||
/// 释放资源 |
|||
func dispose() { |
|||
_ = stopRecording() |
|||
_ = stopPlaying() |
|||
|
|||
audioEngine?.stop() |
|||
audioEngine = nil |
|||
audioPlayer = nil |
|||
audioBufferQueue.removeAll() |
|||
|
|||
os_log("音频管理器已释放", log: log, type: .info) |
|||
} |
|||
|
|||
/// 计算音频缓冲区的RMS值 |
|||
private func calculateRMS(buffer: AVAudioPCMBuffer) -> Float { |
|||
guard let channelData = buffer.int16ChannelData?[0], |
|||
buffer.frameLength > 0 else { |
|||
return 0.0 |
|||
} |
|||
|
|||
let frameCount = Int(buffer.frameLength) |
|||
var rms: Float = 0.0 |
|||
|
|||
// 使用Accelerate框架高效计算RMS |
|||
vDSP_measqv(channelData, 1, &rms, vDSP_Length(frameCount)) |
|||
|
|||
// 计算平方根并归一化 |
|||
rms = sqrt(rms / Float(frameCount)) / 32768.0 |
|||
|
|||
return rms |
|||
} |
|||
} |
|||
@ -0,0 +1,319 @@ |
|||
import Flutter |
|||
import UIKit |
|||
import AVFoundation |
|||
import os.log |
|||
|
|||
/** |
|||
* 实时语音聊天插件 |
|||
* |
|||
* 通过WebSocket连接Vocode服务器实现实时语音交互 |
|||
* 包含录音、播放、WebSocket通信等功能 |
|||
*/ |
|||
@objc public class RealtimePlugin: NSObject, FlutterPlugin { |
|||
|
|||
private let log = OSLog(subsystem: "com.yunqiinnovation.realtime", category: "RealtimePlugin") |
|||
|
|||
// Flutter通道 |
|||
private var methodChannel: FlutterMethodChannel? |
|||
private var eventChannel: FlutterEventChannel? |
|||
private var eventSink: FlutterEventSink? |
|||
|
|||
// 核心组件 |
|||
private let audioManager = RealtimeAudioManager() |
|||
private let websocketManager = RealtimeWebSocketManager() |
|||
|
|||
// 配置参数 |
|||
private var serverUrl: String = "" |
|||
private var sampleRate: Int = 16000 |
|||
private var channels: Int = 1 |
|||
private var bitsPerSample: Int = 16 |
|||
|
|||
// 状态 |
|||
private var connectionStatus: ConnectionStatus = .disconnected |
|||
private var voiceStatus: VoiceStatus = .idle |
|||
|
|||
/// 插件注册 |
|||
public static func register(with registrar: FlutterPluginRegistrar) { |
|||
let instance = RealtimePlugin() |
|||
|
|||
// 方法通道 |
|||
let methodChannel = FlutterMethodChannel(name: "realtime/methods", binaryMessenger: registrar.messenger()) |
|||
registrar.addMethodCallDelegate(instance, channel: methodChannel) |
|||
instance.methodChannel = methodChannel |
|||
|
|||
// 事件通道 |
|||
let eventChannel = FlutterEventChannel(name: "realtime/events", binaryMessenger: registrar.messenger()) |
|||
eventChannel.setStreamHandler(instance) |
|||
instance.eventChannel = eventChannel |
|||
|
|||
// 设置组件回调 |
|||
instance.setupCallbacks() |
|||
} |
|||
|
|||
/// 设置各组件的回调 |
|||
private func setupCallbacks() { |
|||
// 音频管理器回调 |
|||
audioManager.onAudioData = { [weak self] data in |
|||
// 发送到WebSocket |
|||
self?.websocketManager.sendAudioData(data) |
|||
// 同时发送PCM数据到Flutter用于可视化 |
|||
self?.sendPCMDataToFlutter(data) |
|||
} |
|||
|
|||
audioManager.onStatusChanged = { [weak self] status in |
|||
self?.voiceStatus = status |
|||
self?.sendEvent(type: "voiceStatusChanged", data: status.rawValue) |
|||
} |
|||
|
|||
// 新增RMS回调 |
|||
audioManager.onRmsChanged = { [weak self] rms in |
|||
self?.sendEvent(type: "audioRms", data: rms) |
|||
} |
|||
|
|||
// WebSocket管理器回调 |
|||
websocketManager.onConnectionStatusChanged = { [weak self] status in |
|||
self?.connectionStatus = status |
|||
self?.sendEvent(type: "connectionStatusChanged", data: status.rawValue) |
|||
} |
|||
|
|||
websocketManager.onAudioReceived = { [weak self] data in |
|||
// 添加音频数据前50字节的日志输出 |
|||
let dataCount = min(data.count, 50) |
|||
let dataPreview = data.prefix(dataCount).map { String(format: "%02x", $0) }.joined(separator: " ") |
|||
let previewText = data.count > 50 ? dataPreview + "..." : dataPreview |
|||
self?.os_log("收到音频数据前50字节: %@", log: self?.log ?? OSLog.default, type: .debug, previewText) |
|||
self?.os_log("音频数据总长度: %d 字节", log: self?.log ?? OSLog.default, type: .debug, data.count) |
|||
|
|||
self?.audioManager.playAudioData(data) |
|||
// 同时发送接收到的PCM数据到Flutter用于可视化 |
|||
self?.sendPCMDataToFlutter(data) |
|||
} |
|||
|
|||
websocketManager.onTextReceived = { [weak self] text in |
|||
// 直接发送原始JSON文本到Flutter层处理 |
|||
self?.sendEvent(type: "textReceived", data: text) |
|||
} |
|||
|
|||
websocketManager.onError = { [weak self] error in |
|||
self?.sendEvent(type: "error", data: error) |
|||
} |
|||
} |
|||
|
|||
/// 发送事件到Flutter端 |
|||
private func sendEvent(type: String, data: Any?) { |
|||
guard let eventSink = eventSink else { return } |
|||
|
|||
let event: [String: Any] = [ |
|||
"type": type, |
|||
"data": data ?? NSNull() |
|||
] |
|||
|
|||
DispatchQueue.main.async { |
|||
eventSink(event) |
|||
} |
|||
} |
|||
|
|||
/// 将PCM数据发送到Flutter用于可视化 |
|||
private func sendPCMDataToFlutter(_ data: Data) { |
|||
// 确保数据长度是偶数(每2个字节一个16位采样) |
|||
let validSize = data.count & ~1 // 清除最低位,确保偶数 |
|||
|
|||
guard validSize > 0 else { return } |
|||
|
|||
// 将Data转换为Int16数组 (PCM16格式) |
|||
let pcm16Array = data.prefix(validSize).withUnsafeBytes { bytes in |
|||
let int16Pointer = bytes.bindMemory(to: Int16.self) |
|||
return Array(UnsafeBufferPointer(start: int16Pointer.baseAddress, count: validSize / 2)) |
|||
} |
|||
|
|||
// 转换为Int数组以便传递给Flutter |
|||
let pcmData = pcm16Array.map { Int($0) } |
|||
sendEvent(type: "pcmData", data: pcmData) |
|||
} |
|||
|
|||
/// 辅助日志方法 |
|||
private func os_log(_ message: String, log: OSLog, type: OSLogType, _ args: CVarArg...) { |
|||
if args.isEmpty { |
|||
os_log("%@", log: log, type: type, message) |
|||
} else { |
|||
os_log(message, log: log, type: type, args) |
|||
} |
|||
} |
|||
} |
|||
|
|||
// MARK: - FlutterPlugin |
|||
extension RealtimePlugin { |
|||
|
|||
public func handle(_ call: FlutterMethodCall, result: @escaping FlutterResult) { |
|||
switch call.method { |
|||
case "initialize": |
|||
handleInitialize(call, result) |
|||
case "connect": |
|||
handleConnect(result) |
|||
case "disconnect": |
|||
handleDisconnect(result) |
|||
case "startRecording": |
|||
handleStartRecording(result) |
|||
case "stopRecording": |
|||
handleStopRecording(result) |
|||
case "stopPlaying": |
|||
handleStopPlaying(result) |
|||
case "getConnectionStatus": |
|||
result(connectionStatus.rawValue) |
|||
case "getVoiceStatus": |
|||
result(voiceStatus.rawValue) |
|||
case "sendTextMessage": |
|||
handleSendTextMessage(call, result) |
|||
case "setAudioConfig": |
|||
handleSetAudioConfig(call, result) |
|||
case "dispose": |
|||
handleDispose(result) |
|||
default: |
|||
result(FlutterMethodNotImplemented) |
|||
} |
|||
} |
|||
|
|||
private func handleInitialize(_ call: FlutterMethodCall, _ result: @escaping FlutterResult) { |
|||
guard let args = call.arguments as? [String: Any], |
|||
let serverUrl = args["serverUrl"] as? String else { |
|||
result(FlutterError(code: "INVALID_ARGUMENTS", message: "服务器地址不能为空", details: nil)) |
|||
return |
|||
} |
|||
|
|||
self.serverUrl = serverUrl |
|||
self.sampleRate = args["sampleRate"] as? Int ?? 16000 |
|||
self.channels = args["channels"] as? Int ?? 1 |
|||
self.bitsPerSample = args["bitsPerSample"] as? Int ?? 16 |
|||
|
|||
// 初始化音频管理器 |
|||
let audioConfig = AudioConfig( |
|||
sampleRate: sampleRate, |
|||
channels: channels, |
|||
bitsPerSample: bitsPerSample |
|||
) |
|||
|
|||
let success = audioManager.initialize(config: audioConfig) |
|||
if success { |
|||
websocketManager.initialize(serverUrl: serverUrl) |
|||
os_log("实时语音插件初始化成功", log: log, type: .info) |
|||
} else { |
|||
os_log("实时语音插件初始化失败", log: log, type: .error) |
|||
} |
|||
|
|||
result(success) |
|||
} |
|||
|
|||
private func handleConnect(_ result: @escaping FlutterResult) { |
|||
websocketManager.connect { [weak self] success in |
|||
DispatchQueue.main.async { |
|||
result(success) |
|||
} |
|||
} |
|||
} |
|||
|
|||
private func handleDisconnect(_ result: @escaping FlutterResult) { |
|||
websocketManager.disconnect() |
|||
result(true) |
|||
} |
|||
|
|||
private func handleStartRecording(_ result: @escaping FlutterResult) { |
|||
let success = audioManager.startRecording() |
|||
result(success) |
|||
} |
|||
|
|||
private func handleStopRecording(_ result: @escaping FlutterResult) { |
|||
let success = audioManager.stopRecording() |
|||
result(success) |
|||
} |
|||
|
|||
private func handleStopPlaying(_ result: @escaping FlutterResult) { |
|||
let success = audioManager.stopPlaying() |
|||
result(success) |
|||
} |
|||
|
|||
private func handleSendTextMessage(_ call: FlutterMethodCall, _ result: @escaping FlutterResult) { |
|||
guard let args = call.arguments as? [String: Any], |
|||
let message = args["message"] as? String else { |
|||
result(FlutterError(code: "INVALID_ARGUMENTS", message: "消息内容不能为空", details: nil)) |
|||
return |
|||
} |
|||
|
|||
let success = websocketManager.sendTextMessage(message) |
|||
result(success) |
|||
} |
|||
|
|||
private func handleSetAudioConfig(_ call: FlutterMethodCall, _ result: @escaping FlutterResult) { |
|||
guard let args = call.arguments as? [String: Any] else { |
|||
result(FlutterError(code: "INVALID_ARGUMENTS", message: "参数无效", details: nil)) |
|||
return |
|||
} |
|||
|
|||
if let sampleRate = args["sampleRate"] as? Int { |
|||
self.sampleRate = sampleRate |
|||
} |
|||
if let channels = args["channels"] as? Int { |
|||
self.channels = channels |
|||
} |
|||
if let bitsPerSample = args["bitsPerSample"] as? Int { |
|||
self.bitsPerSample = bitsPerSample |
|||
} |
|||
|
|||
let audioConfig = AudioConfig( |
|||
sampleRate: sampleRate, |
|||
channels: channels, |
|||
bitsPerSample: bitsPerSample |
|||
) |
|||
|
|||
let success = audioManager.updateConfig(config: audioConfig) |
|||
result(success) |
|||
} |
|||
|
|||
private func handleDispose(_ result: @escaping FlutterResult) { |
|||
audioManager.dispose() |
|||
websocketManager.dispose() |
|||
result(nil) |
|||
} |
|||
} |
|||
|
|||
// MARK: - FlutterStreamHandler |
|||
extension RealtimePlugin: FlutterStreamHandler { |
|||
|
|||
public func onListen(withArguments arguments: Any?, eventSink events: @escaping FlutterEventSink) -> FlutterError? { |
|||
self.eventSink = events |
|||
return nil |
|||
} |
|||
|
|||
public func onCancel(withArguments arguments: Any?) -> FlutterError? { |
|||
self.eventSink = nil |
|||
return nil |
|||
} |
|||
} |
|||
|
|||
// MARK: - 枚举定义 |
|||
|
|||
enum ConnectionStatus: String { |
|||
case disconnected = "disconnected" |
|||
case connecting = "connecting" |
|||
case connected = "connected" |
|||
case error = "error" |
|||
} |
|||
|
|||
enum VoiceStatus: String { |
|||
case idle = "idle" |
|||
case recording = "recording" |
|||
case processing = "processing" |
|||
case playing = "playing" |
|||
} |
|||
|
|||
// MARK: - 音频配置 |
|||
|
|||
struct AudioConfig { |
|||
let sampleRate: Int |
|||
let channels: Int |
|||
let bitsPerSample: Int |
|||
|
|||
var frameSize: Int { |
|||
// 20ms的帧大小 |
|||
return (sampleRate * channels * bitsPerSample / 8 * 20) / 1000 |
|||
} |
|||
} |
|||
@ -0,0 +1,323 @@ |
|||
import Foundation |
|||
import Network |
|||
import os.log |
|||
|
|||
/** |
|||
* 实时WebSocket管理器 |
|||
* |
|||
* 负责与Vocode服务器的WebSocket通信,对应Android示例中的OkHttp WebSocket功能 |
|||
*/ |
|||
class RealtimeWebSocketManager: NSObject { |
|||
|
|||
private let log = OSLog(subsystem: "com.yunqiinnovation.realtime", category: "RealtimeWebSocketManager") |
|||
|
|||
// WebSocket连接 |
|||
private var webSocketTask: URLSessionWebSocketTask? |
|||
private var urlSession: URLSession? |
|||
|
|||
// 服务器URL |
|||
private var serverUrl: String = "" |
|||
|
|||
// 连接状态 |
|||
private var isConnected = false |
|||
private var isConnecting = false |
|||
|
|||
// 回调 |
|||
var onConnectionStatusChanged: ((ConnectionStatus) -> Void)? |
|||
var onAudioReceived: ((Data) -> Void)? |
|||
var onTextReceived: ((String) -> Void)? |
|||
var onError: ((String) -> Void)? |
|||
|
|||
// 发送队列 |
|||
private let sendQueue = DispatchQueue(label: "com.realtime.websocket.send", qos: .userInitiated) |
|||
private let receiveQueue = DispatchQueue(label: "com.realtime.websocket.receive", qos: .userInitiated) |
|||
|
|||
override init() { |
|||
super.init() |
|||
setupURLSession() |
|||
} |
|||
|
|||
/// 设置URL会话 |
|||
private func setupURLSession() { |
|||
let config = URLSessionConfiguration.default |
|||
config.timeoutIntervalForRequest = 30 |
|||
config.timeoutIntervalForResource = 60 |
|||
urlSession = URLSession(configuration: config, delegate: self, delegateQueue: nil) |
|||
} |
|||
|
|||
/// 初始化WebSocket管理器 |
|||
func initialize(serverUrl: String) { |
|||
self.serverUrl = serverUrl |
|||
os_log("WebSocket管理器初始化,服务器地址: %@", log: log, type: .info, serverUrl) |
|||
} |
|||
|
|||
/// 连接到服务器 |
|||
func connect(completion: @escaping (Bool) -> Void) { |
|||
guard !isConnected && !isConnecting else { |
|||
completion(false) |
|||
return |
|||
} |
|||
|
|||
guard let url = URL(string: serverUrl) else { |
|||
os_log("无效的服务器地址: %@", log: log, type: .error, serverUrl) |
|||
onError?("无效的服务器地址") |
|||
completion(false) |
|||
return |
|||
} |
|||
|
|||
isConnecting = true |
|||
onConnectionStatusChanged?(.connecting) |
|||
|
|||
// 创建WebSocket任务 |
|||
var request = URLRequest(url: url) |
|||
request.timeoutInterval = 30 |
|||
|
|||
webSocketTask = urlSession?.webSocketTask(with: request) |
|||
|
|||
// 开始监听消息 |
|||
startListening() |
|||
|
|||
// 开始连接 |
|||
webSocketTask?.resume() |
|||
|
|||
// 等待连接建立 |
|||
DispatchQueue.global().asyncAfter(deadline: .now() + 1.0) { [weak self] in |
|||
self?.checkConnection(completion: completion) |
|||
} |
|||
} |
|||
|
|||
/// 检查连接状态 |
|||
private func checkConnection(completion: @escaping (Bool) -> Void) { |
|||
guard let webSocketTask = webSocketTask else { |
|||
isConnecting = false |
|||
onConnectionStatusChanged?(.error) |
|||
completion(false) |
|||
return |
|||
} |
|||
|
|||
switch webSocketTask.state { |
|||
case .running: |
|||
isConnected = true |
|||
isConnecting = false |
|||
onConnectionStatusChanged?(.connected) |
|||
os_log("WebSocket连接成功", log: log, type: .info) |
|||
completion(true) |
|||
case .suspended: |
|||
// 等待连接完成 |
|||
DispatchQueue.global().asyncAfter(deadline: .now() + 0.5) { [weak self] in |
|||
self?.checkConnection(completion: completion) |
|||
} |
|||
default: |
|||
isConnecting = false |
|||
onConnectionStatusChanged?(.error) |
|||
os_log("WebSocket连接失败", log: log, type: .error) |
|||
completion(false) |
|||
} |
|||
} |
|||
|
|||
/// 开始监听消息 |
|||
private func startListening() { |
|||
receiveMessage() |
|||
} |
|||
|
|||
/// 接收消息 |
|||
private func receiveMessage() { |
|||
webSocketTask?.receive { [weak self] result in |
|||
switch result { |
|||
case .success(let message): |
|||
self?.handleReceivedMessage(message) |
|||
// 继续监听下一条消息 |
|||
self?.receiveMessage() |
|||
case .failure(let error): |
|||
self?.handleReceiveError(error) |
|||
} |
|||
} |
|||
} |
|||
|
|||
/// 处理接收到的消息 |
|||
private func handleReceivedMessage(_ message: URLSessionWebSocketTask.Message) { |
|||
switch message { |
|||
case .data(let data): |
|||
// 保留对二进制数据的支持以防万一 |
|||
onAudioReceived?(data) |
|||
case .string(let text): |
|||
// 添加收到文本消息的日志输出,显示前20字符 |
|||
let textPreview = text.count > 20 ? String(text.prefix(20)) + "..." : text |
|||
os_log("收到文本消息前20字符: %@", log: log, type: .debug, textPreview) |
|||
|
|||
// 尝试解析为JSON来检查消息类型 |
|||
do { |
|||
if let jsonData = text.data(using: .utf8), |
|||
let json = try JSONSerialization.jsonObject(with: jsonData) as? [String: Any], |
|||
let messageType = json["type"] as? String { |
|||
|
|||
// 添加消息类型的日志输出 |
|||
os_log("消息类型: %@", log: log, type: .debug, messageType) |
|||
|
|||
switch messageType { |
|||
case "websocket_audio": |
|||
// 处理音频消息 |
|||
if let audioDataString = json["data"] as? String, |
|||
let audioData = Data(base64Encoded: audioDataString) { |
|||
onAudioReceived?(audioData) |
|||
} |
|||
case "websocket_transcript": |
|||
// 处理统一的文本消息格式 |
|||
// 包含sender、is_final、timestamp等字段 |
|||
onTextReceived?(text) |
|||
case "websocket_ready": |
|||
// 连接就绪消息 |
|||
os_log("WebSocket连接就绪", log: log, type: .info) |
|||
case "websocket_stop": |
|||
// 会话停止消息 |
|||
os_log("会话已结束", log: log, type: .info) |
|||
case "websocket_start": |
|||
// 会话开始消息 |
|||
os_log("会话已开始", log: log, type: .info) |
|||
default: |
|||
// 其他消息类型,当作普通文本消息处理 |
|||
onTextReceived?(text) |
|||
} |
|||
} else { |
|||
// 如果不是JSON格式,当作普通文本消息处理 |
|||
os_log("非JSON格式消息,当作普通文本处理", log: log, type: .debug) |
|||
onTextReceived?(text) |
|||
} |
|||
} catch { |
|||
// 如果解析失败,当作普通文本消息处理 |
|||
os_log("非JSON格式消息,当作普通文本处理", log: log, type: .debug) |
|||
onTextReceived?(text) |
|||
} |
|||
@unknown default: |
|||
os_log("收到未知类型消息", log: log, type: .error) |
|||
} |
|||
} |
|||
|
|||
/// 处理接收错误 |
|||
private func handleReceiveError(_ error: Error) { |
|||
os_log("WebSocket接收错误: %@", log: log, type: .error, error.localizedDescription) |
|||
|
|||
// 如果连接断开,尝试重连 |
|||
if isConnected { |
|||
isConnected = false |
|||
onConnectionStatusChanged?(.error) |
|||
onError?("连接断开: \(error.localizedDescription)") |
|||
} |
|||
} |
|||
|
|||
/// 发送音频数据 |
|||
func sendAudioData(_ data: Data) { |
|||
guard isConnected else { return } |
|||
|
|||
sendQueue.async { [weak self] in |
|||
// 将音频数据编码为Base64并包装成Vocode WebSocket格式 |
|||
let base64AudioData = data.base64EncodedString() |
|||
let audioMessage: [String: Any] = [ |
|||
"type": "websocket_audio", |
|||
"data": base64AudioData |
|||
] |
|||
|
|||
do { |
|||
let jsonData = try JSONSerialization.data(withJSONObject: audioMessage) |
|||
if let jsonString = String(data: jsonData, encoding: .utf8) { |
|||
self?.webSocketTask?.send(.string(jsonString)) { error in |
|||
if let error = error { |
|||
self?.os_log("发送音频数据失败: %@", log: self?.log ?? OSLog.default, type: .error, error.localizedDescription) |
|||
} |
|||
} |
|||
} |
|||
} catch { |
|||
self?.os_log("音频数据JSON序列化失败: %@", log: self?.log ?? OSLog.default, type: .error, error.localizedDescription) |
|||
} |
|||
} |
|||
} |
|||
|
|||
/// 发送文本消息 |
|||
func sendTextMessage(_ message: String) -> Bool { |
|||
guard isConnected else { |
|||
os_log("WebSocket未连接,无法发送文本消息", log: log, type: .error) |
|||
return false |
|||
} |
|||
|
|||
sendQueue.async { [weak self] in |
|||
self?.webSocketTask?.send(.string(message)) { error in |
|||
if let error = error { |
|||
self?.os_log("发送文本消息失败: %@", log: self?.log ?? OSLog.default, type: .error, error.localizedDescription) |
|||
} |
|||
} |
|||
} |
|||
|
|||
return true |
|||
} |
|||
|
|||
/// 断开连接 |
|||
func disconnect() { |
|||
guard isConnected || isConnecting else { return } |
|||
|
|||
webSocketTask?.cancel(with: .goingAway, reason: nil) |
|||
webSocketTask = nil |
|||
|
|||
isConnected = false |
|||
isConnecting = false |
|||
onConnectionStatusChanged?(.disconnected) |
|||
os_log("WebSocket已断开连接", log: log, type: .info) |
|||
} |
|||
|
|||
/// 释放资源 |
|||
func dispose() { |
|||
disconnect() |
|||
urlSession?.invalidateAndCancel() |
|||
urlSession = nil |
|||
os_log("WebSocket管理器已释放", log: log, type: .info) |
|||
} |
|||
|
|||
// 修复方法访问性问题 |
|||
private func os_log(_ message: String, log: OSLog, type: OSLogType, _ args: CVarArg...) { |
|||
if args.isEmpty { |
|||
os_log("%@", log: log, type: type, message) |
|||
} else { |
|||
os_log(message, log: log, type: type, args) |
|||
} |
|||
} |
|||
} |
|||
|
|||
// MARK: - URLSessionWebSocketDelegate |
|||
extension RealtimeWebSocketManager: URLSessionWebSocketDelegate { |
|||
|
|||
func urlSession(_ session: URLSession, webSocketTask: URLSessionWebSocketTask, didOpenWithProtocol protocol: String?) { |
|||
os_log("WebSocket连接已建立", log: log, type: .info) |
|||
isConnected = true |
|||
isConnecting = false |
|||
onConnectionStatusChanged?(.connected) |
|||
} |
|||
|
|||
func urlSession(_ session: URLSession, webSocketTask: URLSessionWebSocketTask, didCloseWith closeCode: URLSessionWebSocketTask.CloseCode, reason: Data?) { |
|||
os_log("WebSocket连接已关闭,关闭码: %d", log: log, type: .info, closeCode.rawValue) |
|||
isConnected = false |
|||
isConnecting = false |
|||
onConnectionStatusChanged?(.disconnected) |
|||
} |
|||
} |
|||
|
|||
// MARK: - URLSessionDelegate |
|||
extension RealtimeWebSocketManager: URLSessionDelegate { |
|||
|
|||
func urlSession(_ session: URLSession, didBecomeInvalidWithError error: Error?) { |
|||
if let error = error { |
|||
os_log("URLSession失效: %@", log: log, type: .error, error.localizedDescription) |
|||
onError?("会话失效: \(error.localizedDescription)") |
|||
} |
|||
} |
|||
|
|||
func urlSession(_ session: URLSession, task: URLSessionTask, didCompleteWithError error: Error?) { |
|||
if let error = error { |
|||
os_log("URLSessionTask完成时出错: %@", log: log, type: .error, error.localizedDescription) |
|||
if isConnected || isConnecting { |
|||
isConnected = false |
|||
isConnecting = false |
|||
onConnectionStatusChanged?(.error) |
|||
onError?("连接错误: \(error.localizedDescription)") |
|||
} |
|||
} |
|||
} |
|||
} |
|||
@ -0,0 +1,243 @@ |
|||
import 'dart:async'; |
|||
import 'dart:typed_data'; |
|||
|
|||
import 'package:flutter/services.dart'; |
|||
|
|||
/// 实时语音服务异常 |
|||
class RealtimeException implements Exception { |
|||
final String message; |
|||
|
|||
RealtimeException(this.message); |
|||
|
|||
@override |
|||
String toString() => 'RealtimeException: $message'; |
|||
} |
|||
|
|||
/// 连接状态 |
|||
enum ConnectionStatus { |
|||
disconnected, |
|||
connecting, |
|||
connected, |
|||
error, |
|||
} |
|||
|
|||
/// 语音状态 |
|||
enum VoiceStatus { |
|||
idle, |
|||
recording, |
|||
processing, |
|||
playing, |
|||
} |
|||
|
|||
/// 实时语音事件 |
|||
class RealtimeEvent { |
|||
final String type; |
|||
final dynamic data; |
|||
|
|||
RealtimeEvent({required this.type, this.data}); |
|||
|
|||
factory RealtimeEvent.fromMap(Map<String, dynamic> map) { |
|||
return RealtimeEvent( |
|||
type: map['type'] as String, |
|||
data: map['data'], |
|||
); |
|||
} |
|||
} |
|||
|
|||
/// 实时语音聊天服务 |
|||
class RealtimeService { |
|||
static const MethodChannel _channel = MethodChannel('realtime/methods'); |
|||
static const EventChannel _eventChannel = EventChannel('realtime/events'); |
|||
|
|||
StreamController<RealtimeEvent>? _eventStreamController; |
|||
Stream<RealtimeEvent>? _eventStream; |
|||
|
|||
/// 获取事件流 |
|||
Stream<RealtimeEvent> get eventStream { |
|||
if (_eventStream == null) { |
|||
_eventStreamController = StreamController<RealtimeEvent>.broadcast(); |
|||
_eventStream = _eventStreamController!.stream; |
|||
|
|||
// 监听原生事件 |
|||
_eventChannel.receiveBroadcastStream().listen( |
|||
(dynamic event) { |
|||
if (event is Map<dynamic, dynamic>) { |
|||
final eventMap = Map<String, dynamic>.from(event); |
|||
final realtimeEvent = RealtimeEvent.fromMap(eventMap); |
|||
_eventStreamController!.add(realtimeEvent); |
|||
} |
|||
}, |
|||
onError: (error) { |
|||
_eventStreamController!.addError(RealtimeException('事件流错误: $error')); |
|||
}, |
|||
); |
|||
} |
|||
|
|||
return _eventStream!; |
|||
} |
|||
|
|||
/// 初始化实时语音服务 |
|||
/// |
|||
/// [serverUrl] WebSocket服务器地址 |
|||
/// [sampleRate] 采样率,默认16000 |
|||
/// [channels] 声道数,默认1(单声道) |
|||
/// [bitsPerSample] 位深,默认16 |
|||
Future<bool> initialize({ |
|||
required String serverUrl, |
|||
int sampleRate = 16000, |
|||
int channels = 1, |
|||
int bitsPerSample = 16, |
|||
}) async { |
|||
try { |
|||
final result = await _channel.invokeMethod<bool>( |
|||
'initialize', |
|||
{ |
|||
'serverUrl': serverUrl, |
|||
'sampleRate': sampleRate, |
|||
'channels': channels, |
|||
'bitsPerSample': bitsPerSample, |
|||
}, |
|||
); |
|||
|
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('初始化失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 连接到服务器 |
|||
Future<bool> connect() async { |
|||
try { |
|||
final result = await _channel.invokeMethod<bool>('connect'); |
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('连接失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 断开连接 |
|||
Future<bool> disconnect() async { |
|||
try { |
|||
final result = await _channel.invokeMethod<bool>('disconnect'); |
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('断开连接失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 开始录音 |
|||
Future<bool> startRecording() async { |
|||
try { |
|||
final result = await _channel.invokeMethod<bool>('startRecording'); |
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('开始录音失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 停止录音 |
|||
Future<bool> stopRecording() async { |
|||
try { |
|||
final result = await _channel.invokeMethod<bool>('stopRecording'); |
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('停止录音失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 停止播放 |
|||
Future<bool> stopPlaying() async { |
|||
try { |
|||
final result = await _channel.invokeMethod<bool>('stopPlaying'); |
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('停止播放失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 获取连接状态 |
|||
Future<ConnectionStatus> getConnectionStatus() async { |
|||
try { |
|||
final result = await _channel.invokeMethod<String>('getConnectionStatus'); |
|||
switch (result) { |
|||
case 'disconnected': |
|||
return ConnectionStatus.disconnected; |
|||
case 'connecting': |
|||
return ConnectionStatus.connecting; |
|||
case 'connected': |
|||
return ConnectionStatus.connected; |
|||
case 'error': |
|||
return ConnectionStatus.error; |
|||
default: |
|||
return ConnectionStatus.disconnected; |
|||
} |
|||
} catch (e) { |
|||
throw RealtimeException('获取连接状态失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 获取语音状态 |
|||
Future<VoiceStatus> getVoiceStatus() async { |
|||
try { |
|||
final result = await _channel.invokeMethod<String>('getVoiceStatus'); |
|||
switch (result) { |
|||
case 'idle': |
|||
return VoiceStatus.idle; |
|||
case 'recording': |
|||
return VoiceStatus.recording; |
|||
case 'processing': |
|||
return VoiceStatus.processing; |
|||
case 'playing': |
|||
return VoiceStatus.playing; |
|||
default: |
|||
return VoiceStatus.idle; |
|||
} |
|||
} catch (e) { |
|||
throw RealtimeException('获取语音状态失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 发送文本消息 |
|||
Future<bool> sendTextMessage(String message) async { |
|||
try { |
|||
final result = await _channel.invokeMethod<bool>( |
|||
'sendTextMessage', |
|||
{'message': message}, |
|||
); |
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('发送文本消息失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 设置音频参数 |
|||
Future<bool> setAudioConfig({ |
|||
int? sampleRate, |
|||
int? channels, |
|||
int? bitsPerSample, |
|||
}) async { |
|||
try { |
|||
final params = <String, dynamic>{}; |
|||
if (sampleRate != null) params['sampleRate'] = sampleRate; |
|||
if (channels != null) params['channels'] = channels; |
|||
if (bitsPerSample != null) params['bitsPerSample'] = bitsPerSample; |
|||
|
|||
final result = await _channel.invokeMethod<bool>('setAudioConfig', params); |
|||
return result ?? false; |
|||
} catch (e) { |
|||
throw RealtimeException('设置音频参数失败: $e'); |
|||
} |
|||
} |
|||
|
|||
/// 释放资源 |
|||
Future<void> dispose() async { |
|||
try { |
|||
await _channel.invokeMethod<void>('dispose'); |
|||
_eventStreamController?.close(); |
|||
_eventStreamController = null; |
|||
_eventStream = null; |
|||
} catch (e) { |
|||
throw RealtimeException('释放资源失败: $e'); |
|||
} |
|||
} |
|||
} |
|||
@ -0,0 +1,29 @@ |
|||
name: realtime |
|||
description: 实时语音聊天插件,通过WebSocket连接Vocode服务器实现语音交互 |
|||
version: 0.0.1 |
|||
homepage: |
|||
|
|||
environment: |
|||
sdk: ">=2.17.0 <3.0.0" |
|||
flutter: ">=2.5.0" |
|||
|
|||
dependencies: |
|||
flutter: |
|||
sdk: flutter |
|||
|
|||
dev_dependencies: |
|||
flutter_test: |
|||
sdk: flutter |
|||
flutter_lints: ^2.0.0 |
|||
|
|||
# Flutter插件配置 |
|||
flutter: |
|||
plugin: |
|||
platforms: |
|||
android: |
|||
package: com.yunqiinnovation.realtime |
|||
pluginClass: RealtimePlugin |
|||
ios: |
|||
pluginClass: RealtimePlugin |
|||
swiftPackage: |
|||
path: ios/realtime |
|||
Loading…
Reference in new issue