加入ios通话翻译功能

SpeexKit.xcframework/Info.plist

@@ -0,0 +1,27 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>AvailableLibraries</key>
+	<array>
+		<dict>
+			<key>BinaryPath</key>
+			<string>SpeexKit.framework/SpeexKit</string>
+			<key>LibraryIdentifier</key>
+			<string>ios-arm64</string>
+			<key>LibraryPath</key>
+			<string>SpeexKit.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>arm64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+		</dict>
+	</array>
+	<key>CFBundlePackageType</key>
+	<string>XFWK</string>
+	<key>XCFrameworkFormatVersion</key>
+	<string>1.0</string>
+</dict>
+</plist>

SpeexKit.xcframework/ios-arm64/SpeexKit.framework/Headers/OpusTools.h

@@ -0,0 +1,44 @@
+//
+//  OpusTools.h
+//  SpeexKit
+//
+//  Created by EzioChan on 2024/1/17.
+//  Copyright © 2024 www.zh-jieli.com. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+
+
+NS_ASSUME_NONNULL_BEGIN
+
+@protocol OpusToolsDelegate <NSObject>
+
+-(void)opusToolsPcmData:(NSData*)data;
+
+-(void)opusToolsPcmFile:(NSData *)data;
+
+-(void)opusToolsOpusData:(NSData *)data;
+
+-(void)opusToolsOpusFile:(NSData *)data;
+
+@end
+
+@interface OpusTools : NSObject
+
+@property(nonatomic,weak)id<OpusToolsDelegate> delegate;
+
+-(void)opusDecodeInfo:(int)sampleRate Channel:(int)channel frameSize:(int) fs isHasHead:(BOOL)head;
+
+-(void)opusEncodeInfo:(int)sampleRate Channel:(int)channel frameSize:(int)fs isHasHead:(BOOL)head;
+
+-(void)decodeData:(NSData*)data;
+
+-(void)decodeFile:(NSString*)path;
+
+-(void)encodeData:(NSData*)data;
+
+-(void)encodeFile:(NSString*)path;
+
+@end
+
+NS_ASSUME_NONNULL_END

SpeexKit.xcframework/ios-arm64/SpeexKit.framework/Headers/OpusUnit.h

@@ -0,0 +1,100 @@
+//
+//  OpusUnit.h
+//  QCY_Demo
+//
+//  Created by 杰理科技 on 2021/7/5.
+//  Copyright © 2021 杰理科技. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ *  通知：流式编码回调Opus数据
+ *  通知数据类型：NSData（Opus）
+ */
+extern NSString *kOPUS_ENCODE_DATA;
+/**
+ *  通知：流式解码回调PCM数据
+ *  通知数据类型：NSData（PCM）
+ */
+extern NSString *kOPUS_DECODE_DATA;
+
+
+@interface OpusUnit : NSObject
+
++ (void)opusIsLog:(BOOL)log;
+
+#pragma mark - Opus参数
+/**
+ *  设置参数
+ *  @param rate    采样率 例如：48000、24000、16000、12000、8000（默认 16000）
+ *  @param frameRate 帧长度
+ *  16k ：40
+ *  32K ：80
+ *  64k ：160
+ *  96k ：320
+ *  128k：640
+ *  @param channel    声道 例如：1、2 （默认 1）
+ *  @param isHeader    是否带头
+ */
++(void)opusSetSampleRate:(int)rate frameRate:(int)frameRate Channels:(int)channel isHeader:(BOOL)isHeader;
+
+#pragma mark - Opus解码
+
+/**
+ *  直接【opus文件】转换成【pcm文件】
+ *  @param path_opus    opus文件路径
+ *  @param path_pcm    pcm文件路径
+ */
++ (int)opusDecodeOPUS:(NSString *)path_opus PCM:(NSString *)path_pcm;
+
+/**
+ *  流式解码【开启】
+ *  注意：需要异步调用，放在工程单利类中，切勿频繁开启。
+ */
++ (int)opusDecoderRun;
+
+/**
+ *  输入Opus的数据
+ *  通知监听“kOPUS_DECODE_DATA”获得解码后数据
+ *  通知数据类型：NSNotification.object => NSData（PCM格式）
+ *  @param data    opus数据流（长度：1024）
+ */
++ (void)opusWriteData:(NSData*)data;
+
+/**
+ *  流式解码【关闭】
+ */
++ (int)opusDecoderStop;
+
+#pragma mark - Opus编码
+
+/**
+ *  【pcm文件】转换成【opus文件】
+ */
++ (int)opusEncodePCM:(NSString *)path_pcm OPUS:(NSString *)path_opus;
+
+/**
+ *  流式编码【开启】
+ *  注意：需要异步调用，放在工程单利类中，切勿频繁开启。
+ */
++ (int)opusEncoderRun;
+
+/**
+ *  输入pcm的数据
+ *  通知监听“kOPUS_DECODE_DATA”获得解码后数据
+ *  通知数据类型：NSNotification.object =>  NSData（opus格式）
+ *  @param data    opus数据流（长度：1024）
+ */
++ (void)pcmWriteData:(NSData*)data;
+
+/**
+ *  流式编码【关闭】
+ */
++ (int)opusEncoderStop;
+
+@end
+
+NS_ASSUME_NONNULL_END

SpeexKit.xcframework/ios-arm64/SpeexKit.framework/Headers/SpeexKit.h

@@ -0,0 +1,21 @@
+//
+//  SpeexKit.h
+//  SpeexKit
+//
+//  Created by zhihui liang on 2018/8/20.
+//  Copyright © 2018年 www.zh-jieli.com. All rights reserved.
+//
+
+#import <UIKit/UIKit.h>
+
+//! Project version number for SpeexKit.
+FOUNDATION_EXPORT double SpeexKitVersionNumber;
+
+//! Project version string for SpeexKit.
+FOUNDATION_EXPORT const unsigned char SpeexKitVersionString[];
+
+// In this header, you should import all the public headers of your framework using statements like #import <SpeexKit/PublicHeader.h>
+
+#import <SpeexKit/SpeexUnit.h>
+#import <SpeexKit/OpusUnit.h>
+#import <SpeexKit/OpusTools.h>

SpeexKit.xcframework/ios-arm64/SpeexKit.framework/Headers/SpeexUnit.h

@@ -0,0 +1,40 @@
+//
+//  SpeexUnit.h
+//  SpeexKit
+//
+//  Created by zhihui liang on 2018/8/20.
+//  Copyright © 2018年 www.zh-jieli.com. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+
+/**
+ *  流式解码回调PCM数据
+ */
+extern NSString *kSPEEX_DECODE_DATA;       //speex decode
+
+@interface SpeexUnit : NSObject
+
++(void)speexIsLog:(BOOL)log;
+
+/**
+ *  直接【speex文件】转换成【pcm文件】
+ */
++(int)speexDecodeSPX:(NSString *)path_spx PCM:(NSString *)path_pcm;
+
+/**
+ *  流式解码【开启】
+ */
++(int)speexDecoderRun;
+
+/**
+ *  输入Speex的数据
+ */
++(void)speexWriteData:(NSData*)data;
+
+/**
+ *  流式解码【关闭】
+ */
++(int)speexDecoderStop;
+
+@end

SpeexKit.xcframework/ios-arm64/SpeexKit.framework/Modules/module.modulemap

@@ -0,0 +1,6 @@
+framework module SpeexKit {
+  umbrella header "SpeexKit.h"
+  export *
+
+  module * { export * }
+}

lib/modules/settings/views/settings_view.dart

@@ -515,27 +515,27 @@ class SettingsView extends GetView<SettingsController> {
                   //         ? Colors.white.withOpacity(0.1)
                   //         : Colors.grey[200]),
                   //BLE测试
-                  // _buildSimpleNavigationSetting(
+                  _buildSimpleNavigationSetting(
-                  //   title: 'opus解码测试',
+                    title: 'opus解码测试',
-                  //   subtitle: '测试opus解码',
+                    subtitle: '测试opus解码',
-                  //   icon: Icons.bluetooth_searching,
+                    icon: Icons.bluetooth_searching,
-                  //   iconBgColor: isDarkMode
+                    iconBgColor: isDarkMode
-                  //       ? Colors.green[900]!.withOpacity(0.3)
+                        ? Colors.green[900]!.withOpacity(0.3)
-                  //       : Colors.green[100]!,
+                        : Colors.green[100]!,
-                  //   iconColor:
+                    iconColor:
-                  //       isDarkMode ? Colors.green[300]! : Colors.green[600]!,
+                        isDarkMode ? Colors.green[300]! : Colors.green[600]!,
-                  //   titleColor: isDarkMode ? Colors.white : null,
+                    titleColor: isDarkMode ? Colors.white : null,
-                  //   subtitleColor: isDarkMode ? Colors.white70 : null,
+                    subtitleColor: isDarkMode ? Colors.white70 : null,
-                  //   onTap: () {
+                    onTap: () {
-                  //     Get.toNamed(Routes.opusTest);
+                      Get.toNamed(Routes.opusTest);
-                  //   },
+                    },
-                  //   isDarkMode: isDarkMode,
+                    isDarkMode: isDarkMode,
-                  // ),
+                  ),
-                  // Divider(
+                  Divider(
-                  //     height: 1,
+                      height: 1,
-                  //     color: isDarkMode
+                      color: isDarkMode
-                  //         ? Colors.white.withOpacity(0.1)
+                          ? Colors.white.withOpacity(0.1)
-                  //         : Colors.grey[200]),
+                          : Colors.grey[200]),
 
                   // BLE测试
                   // _buildSimpleNavigationSetting(

local_plugins/agent_service/ios/agent_service/Sources/agent_service/AgentServiceImpl.swift

@@ -1463,7 +1463,7 @@ extension AgentServiceImpl: BleService.Callback {
     }
     
     func onAudioDataReceived1(data: Data) {
-    
+
     }
     
     func onWakeupSignalReceived() {

local_plugins/azure_speech/ios/azure_speech/Package.swift

@@ -9,13 +9,16 @@ let package = Package(
   ],
   dependencies: [
     // 依赖speech接口模块
-    .package(path: "../../../speech/ios/speech")
+    .package(path: "../../../speech/ios/speech"),
+    // 添加对ble_service的依赖
+    .package(path: "../../../ble_service/ios/ble_service")
   ],
   targets: [
     .target(
       name: "azure_speech",
       dependencies: [
         .product(name: "speech", package: "speech"),
+        .product(name: "ble-service", package: "ble_service"),
         "MicrosoftCognitiveServicesSpeech"
       ],
       path: "Sources",  // 修改为包含整个 Sources 目录

local_plugins/azure_speech/ios/azure_speech/Sources/azure_speech/AzureAsrHelper.swift

@@ -3,6 +3,7 @@ import AVFoundation
 import MicrosoftCognitiveServicesSpeech
 import speech
 import os.log
+import Network
 
 // MARK: - 基于微软Azure语音服务的ASR实现
 /**
@@ -21,6 +22,12 @@ public class AzureAsrHelper: NSObject {
     private var recognizer: SPXSpeechRecognizer?
     private var audioConfig: SPXAudioConfiguration?
     
+    // 网络监听相关
+    private var networkMonitor: NWPathMonitor?
+    private var networkQueue: DispatchQueue?
+    private var isNetworkAvailable = true
+    private var continuousCallback: ContinuousRecognizeCallback?
+    
     // 状态管理
     private var _isContinuousRecognitionActive = false
     
@@ -110,20 +117,93 @@ public class AzureAsrHelper: NSObject {
             // 预初始化音频组件
             preInitializeAudioComponents()
             
+            // 初始化网络监听
+            initNetworkMonitoring()
+            
             return true
         } catch {
             os_log("初始化失败: %{public}@", log: log, type: .error, error.localizedDescription)
             return false
         }
     }
-    
+     /**
+             * 初始化网络监听
+             */
+            private func initNetworkMonitoring() {
+                networkQueue = DispatchQueue(label: "NetworkMonitor")
+                networkMonitor = NWPathMonitor()
+                
+                networkMonitor?.pathUpdateHandler = { [weak self] path in
+                    guard let self = self else { return }
+                    
+                    let wasAvailable = self.isNetworkAvailable
+                    self.isNetworkAvailable = (path.status == .satisfied)
+                    
+                    if wasAvailable && !self.isNetworkAvailable {
+                        // 网络断开
+                        self.handleNetworkLost()
+                    } else if !wasAvailable && self.isNetworkAvailable {
+                        // 网络恢复
+                        self.handleNetworkAvailable()
+                    }
+                }
+                
+                if let queue = networkQueue {
+                    networkMonitor?.start(queue: queue)
+                }
+                
+                os_log("网络监听已初始化", log: log, type: .info)
+            }
+            
+            /**
+             * 处理网络可用事件
+             */
+            private func handleNetworkAvailable() {
+                os_log("网络已恢复", log: log, type: .info)
+                
+                DispatchQueue.main.async { [weak self] in
+                   // self?.continuousCallback?.onNetworkRestored()
+                }
+            }
+            
+            /**
+             * 处理网络丢失事件
+             */
+            private func handleNetworkLost() {
+                os_log("网络连接丢失", log: log, type: .error)
+                
+                // 立即停止音频处理
+                stopAudioProcessingImmediately()
+                
+                DispatchQueue.main.async { [weak self] in
+                   // self?.continuousCallback?.onNetworkError("网络连接已断开")
+                }
+            }
+            
+            /**
+             * 立即停止音频处理
+             */
+            private func stopAudioProcessingImmediately() {
+                audioStream?.stopMicrophoneCapture()
+                
+                if _isContinuousRecognitionActive {
+                    do {
+                        try recognizer?.stopContinuousRecognition()
+                        _isContinuousRecognitionActive = false
+                        os_log("因网络问题强制停止识别", log: log, type: .info)
+                    } catch {
+                        _isContinuousRecognitionActive = false
+                        os_log("强制停止识别失败: %@", log: log, type: .error, error.localizedDescription)
+                    }
+                }
+            }
     /**
      * 预初始化音频组件
      * 在初始化阶段就准备好音频流和相关组件，减少首次启动延迟
      */
     private func preInitializeAudioComponents() {
         if audioStream == nil {
-            audioStream = SimpleAudioReceiver(parentHelper: self)
+            audioStream = SimpleAudioReceiver()
             audioStream?.initAudioRecord()
         }
         
@@ -149,7 +229,11 @@ public class AzureAsrHelper: NSObject {
     ) -> Bool {
         guard speechConfig != nil else {
             print("语音服务未初始化")
-            //callback.onError("语音服务未初始化")
+            return false
+        }
+        
+        // 检查网络状态
+        if !checkNetworkStatus() {
             return false
         }
         
@@ -170,6 +254,8 @@ public class AzureAsrHelper: NSObject {
                 }
            return false
             }
+           
+            print("startContinuousRecognition:\(audioSourceType)")
             // 启动音频处理
             audioStream?.startAudioRecord(audioSourceType: audioSourceType == .microphone ? .microphone : .external, audioDataCallback: audioDataCallback)
             
@@ -381,7 +467,7 @@ public class AzureAsrHelper: NSObject {
      * 设置麦克风音频流
      */
     private func setupMicrophoneStream() {
-        audioStream = SimpleAudioReceiver(parentHelper: self)
+        audioStream = SimpleAudioReceiver()
           // 【优化】检查音频配置是否已存在
     if audioConfig == nil, let pushStream = audioStream?.pushAudioStream {
         audioConfig = SPXAudioConfiguration(streamInput: pushStream)
@@ -462,15 +548,19 @@ public class AzureAsrHelper: NSObject {
     /**
      * 设置事件监听器
      */
-    public func setupEventListeners(callback: ContinuousRecognizeCallback) -> Bool{
+    public func setupEventListeners(callback: ContinuousRecognizeCallback) -> Bool {
         print("设置ssssss监听器:${speechConfig}")
         
-        //  重设识别器
+        // 保存回调引用用于网络事件
+        self.continuousCallback = callback
+        
+        // 重设识别器
         if (!setupRecognizer()) {
             return false
         }
-        guard let recognizer = recognizer else { return false}
+        guard let recognizer = recognizer else { return false }
-       // 正在识别事件
+        
+        // 正在识别事件
     recognizer.addRecognizingEventHandler { [weak self] (_, event) in
       guard let self = self else { return }
         let result = event.result
@@ -532,7 +622,12 @@ public class AzureAsrHelper: NSObject {
             
             os_log("识别取消: %{public}@", log: self.log, type: .error, errorDetails)
             
-            callback.onCanceled(reason, errorDetails)
+            // 检查是否为网络相关错误
+            if self.isNetworkRelatedError(errorDetails) {
+                callback.onError("网络连接问题: \(errorDetails)")
+            } else {
+                callback.onCanceled(reason, errorDetails)
+            }
         }
         return true
     }
@@ -575,7 +670,7 @@ public class AzureAsrHelper: NSObject {
         
         if audioStream == nil {
             // 创建外部音频拉流对象
-            audioStream = SimpleAudioReceiver(parentHelper: self)
+            audioStream = SimpleAudioReceiver()
             audioStream?.initAudioRecord()
         }
         
@@ -702,6 +797,43 @@ public class AzureAsrHelper: NSObject {
          */
         func onError(_ error: String)
     }
+    /**
+ * 检查网络状态
+ * @return 网络是否可用
+ */
+private func checkNetworkStatus() -> Bool {
+    guard let monitor = networkMonitor else {
+        os_log("网络监听器未初始化，假设网络可用", log: log)
+        return true
+    }
+    
+    let currentPath = monitor.currentPath
+    let isAvailable = currentPath.status == .satisfied
+    
+    if !isAvailable {
+        os_log("网络不可用，无法启动语音识别", log: log, type: .error)
+        continuousCallback?.onError("网络连接不可用，请检查网络设置")
+    }
+    
+    return isAvailable
+}
+
+/**
+ * 判断是否为网络相关错误
+ * @param errorDetails 错误详情
+ * @return 是否为网络错误
+ */
+private func isNetworkRelatedError(_ errorDetails: String) -> Bool {
+    let networkErrorKeywords = [
+        "network", "connection", "timeout", "unreachable",
+        "网络", "连接", "超时", "不可达",
+        "ConnectionFailure", "NetworkError", "TimeoutError"
+    ]
+    
+    let lowercaseError = errorDetails.lowercased()
+    return networkErrorKeywords.contains { lowercaseError.contains($0.lowercased()) }
+}
+
 }
 
 // //ios    语言检测
@@ -754,3 +886,4 @@ public class AzureAsrHelper: NSObject {
 //             "originalCode": topResult.language
 //         ]
 //     }
+

local_plugins/azure_speech/ios/azure_speech/Sources/azure_speech/AzureSpeechPlugin.swift

@@ -5,6 +5,8 @@ import MicrosoftCognitiveServicesSpeech
 // 自定义语音处理组件，提供音频流处理等功能
 import speech
 import os.log
+// 导入蓝牙服务模块，用于处理蓝牙设备回调
+import ble_service
 
 /**
  * Azure Speech Plugin
@@ -32,6 +34,14 @@ import os.log
     
     // 是否已添加TTS事件监听器
     private var isTtsListenerAdded = false
+    // AST相关
+    private var astChannel: FlutterMethodChannel?
+    private var astEventChannel: FlutterEventChannel?
+    private let azureAstHelper = IntegratedSpeechTranslationService()
+      // 创建事件回调
+    private var astEventCallback: AstEventCallback?
+    // 是否已添加AST事件监听器
+    private var isAstListenerAdded = false
     
     // 当前的连续识别回调
     private var currentAsrCallback: AsrCallbackWrapper?
@@ -39,7 +49,9 @@ import os.log
     // 插件注册
     public static func register(with registrar: FlutterPluginRegistrar) {
         let instance = AzureSpeechPlugin()
-        
+            // 2. 初始化BleService并注册回调  
+        BleService.shared.addCallback(instance)
+
         // 初始化ASR通道
         let asrChannel = FlutterMethodChannel(name: "azure_speech/asr", binaryMessenger: registrar.messenger())
         registrar.addMethodCallDelegate(instance, channel: asrChannel)
@@ -60,6 +72,17 @@ import os.log
             instance?.handleTtsMethodCall(call, result: result)
         }
         
+        
+        // 初始化AST通道
+        let astChannel = FlutterMethodChannel(name: "azure_speech/ast", binaryMessenger: registrar.messenger())
+        registrar.addMethodCallDelegate(instance, channel: astChannel)
+        instance.astChannel = astChannel
+        
+        // 设置AST通道处理器 - 使用实例方法
+        astChannel.setMethodCallHandler { [weak instance] (call, result) in
+            instance?.handleAstMethodCall(call, result: result)
+        }
+        
         // 初始化ASR事件通道
         let asrEventChannel = FlutterEventChannel(name: "azure_speech/asr_events", binaryMessenger: registrar.messenger())
         asrEventChannel.setStreamHandler(instance)
@@ -69,6 +92,14 @@ import os.log
         let ttsEventChannel = FlutterEventChannel(name: "azure_speech/tts_events", binaryMessenger: registrar.messenger())
         ttsEventChannel.setStreamHandler(instance)
         instance.ttsEventChannel = ttsEventChannel
+        
+        // 初始化TTS事件通道
+        let astEventChannel = FlutterEventChannel(name: "azure_speech/asr_events", binaryMessenger: registrar.messenger())
+        astEventChannel.setStreamHandler(instance)
+        instance.astEventChannel = astEventChannel
+
+     
+       
     }
     // 发送ASR事件方法
     internal func sendAsrEvent(_ event: [String: Any]) {
@@ -153,20 +184,21 @@ import os.log
                 return
             }
             
-            // 解析音频源类型
+            // 解析音频源类型参数
-            let audioSourceType: AzureAsrHelper.AudioSourceType
+            guard let args = call.arguments as? [String: Any],
-            if let args = call.arguments as? [String: Any],
+                  let isExternalActive = args["audioSourceType"] as? Bool else {
-               let audioSourceString = args["audioSourceType"] as? String {
+                result(FlutterError(code: "INVALID_ARGUMENTS", message: "必要的参数不能为空", details: nil))
-                switch audioSourceString.lowercased() {
+                return
-                case "external":
-                    audioSourceType = .external
-                default:
-                    audioSourceType = .microphone
-                }
-            } else {
-                audioSourceType = .microphone
             }
             
+            // 根据参数确定音频源类型
+            let audioSourceType = isExternalActive ? 
+                AzureAsrHelper.AudioSourceType.external : 
+                AzureAsrHelper.AudioSourceType.microphone
+
+            print("startContinuousRecognition: \(audioSourceType)")
+            
+            // 启动连续识别
             let success = azureAsrHelper.startContinuousRecognition(
                 audioSourceType: audioSourceType
             )
@@ -184,17 +216,17 @@ import os.log
                 callback: currentAsrCallback!
             )
             result(success)    
-
+            
-            case "disableBluetoothAudio":
+        case "disableBluetoothAudio":
-                azureAsrHelper.disableBluetoothAudio();
+            azureAsrHelper.disableBluetoothAudio();
-                return
+            return
-
+            
-            case "restoreOriginalAudioState":
+        case "restoreOriginalAudioState":
-                azureAsrHelper.restoreOriginalAudioState();
+            azureAsrHelper.restoreOriginalAudioState();
-                return
+            return
-
+            
         case "stopContinuousRecognition":
-       
+            
             let success = azureAsrHelper.stopContinuousRecognition()
             currentAsrCallback = nil
             result(success)
@@ -287,15 +319,15 @@ import os.log
             } catch {
                 result(FlutterError(code: "ENABLERECORD_ERROR", message: error.localizedDescription, details: nil))
             }
-    
+            
-    
+            
             
         case "pauseRecord":
             azureAsrHelper.pauseRecord()
             result(true)
-        // case "resumeRecord":
+            // case "resumeRecord":
-        //     azureAsrHelper.resumeRecord()
+            //     azureAsrHelper.resumeRecord()
-        //     result(true)
+            //     result(true)
             
         case "stopRecord":
             guard let args = call.arguments as? [String: Any],
@@ -311,16 +343,16 @@ import os.log
             }
         case "disableBluetoothAudio":
             print("disableBluetoothAudio")
-             azureAsrHelper.disableBluetoothAudio();
+            azureAsrHelper.disableBluetoothAudio();
-          azureTtsHelper.stop()
+            azureTtsHelper.stop()
             result(true)
             
             
             
         case "restoreOriginalAudioState":
             print("restoreOriginalAudioState")
-             azureAsrHelper.restoreOriginalAudioState();
+            azureAsrHelper.restoreOriginalAudioState();
-             azureTtsHelper.stop()
+            azureTtsHelper.stop()
             result(true)
             
         case "setAudioConfig":
@@ -415,13 +447,13 @@ import os.log
             if (type == 0) {
                 // 默认（如果有耳机选耳机，否则使用系统扬声器）
                 azureAsrHelper.restoreOriginalAudioState();
-               // azureTtsHelper.setAudioOutputDevice()
+                // azureTtsHelper.setAudioOutputDevice()
                 
                 
             } else if (type == 1) {
                 // 强制使用声器
                 azureAsrHelper.disableBluetoothAudio();
-               // azureTtsHelper.setAudioOutputDevice()
+                // azureTtsHelper.setAudioOutputDevice()
                 
             } else if (type == 2) {
                 // 强制使用耳机
@@ -437,60 +469,388 @@ import os.log
             result(FlutterMethodNotImplemented)
         }
     }
+    
+    // MARK: - AST 方法处理
+    private func handleAstMethodCall(_ call: FlutterMethodCall, result: @escaping FlutterResult) {
+        switch call.method {
+        case "enableRecord":
+            guard let args = call.arguments as? [String: Any],
+                  let filePath = args["filePath"] as? String else {
+                result(FlutterError(code: "INVALID_ARGUMENTS", message: "Missing filePath", details: nil))
+                return
+            }
+            
+            do {
+                os_log("音频文件名称为: %@", log: log, type: .info, filePath)
+                try azureAstHelper.enableRecord(filePath: filePath)
+                result(true)
+            } catch {
+                result(FlutterError(code: "ENABLERECORD_ERROR", message: error.localizedDescription, details: nil))
+            }
+            
+        case "startContinuousTranslation":
+            do {
+                os_log("开启翻译", log: log, type: .info)
+                try azureAstHelper.startContinuousTranslation()
+                result(true)
+            } catch {
+                result(FlutterError(code: "START_CONTINUOUS_TRANSLATION_ERROR", message: error.localizedDescription, details: nil))
+            }
+            
+        case "stopContinuousTranslation":
+            do {
+                os_log("停止翻译", log: log, type: .info)
+                try azureAstHelper.stopContinuousTranslation()
+                result(true)
+            } catch {
+                result(FlutterError(code: "STOP_CONTINUOUS_TRANSLATION_ERROR", message: error.localizedDescription, details: nil))
+            }
+            
+        case "dispose":
+            do {
+                os_log("释放AST资源", log: log, type: .info)
+                azureAstHelper.dispose()
+                result(true)
+            } catch {
+                result(FlutterError(code: "DISPOSE_ERROR", message: error.localizedDescription, details: nil))
+            }
+            
+        case "recognizeCallback":
+            os_log("recognizeCallback:", log: log, type: .info)
+            result(true)
+            
+        case "path":
+            guard let args = call.arguments as? [String: Any],
+                  let filePath = args["filePath"] as? String else {
+                result(FlutterError(code: "INVALID_ARGUMENTS", message: "Missing filePath", details: nil))
+                return
+            }
+            
+            let fileURL = URL(fileURLWithPath: filePath)
+            
+            // if FileManager.default.fileExists(atPath: filePath) {
+            //     // 启动异步任务来按播放速度读取音频文件
+            //     Task {
+            //       //  await streamAudioFile(fileURL: fileURL)
+            //     }
+            //     result(true)
+            // } else {
+            //     result(FlutterError(code: "FILE_NOT_FOUND", message: "音频文件不存在: \(filePath)", details: nil))
+            // }
+            
+        case "initialize":
+            guard let args = call.arguments as? [String: Any] else {
+                result(FlutterError(code: "INVALID_ARGUMENTS", message: "Missing arguments", details: nil))
+                return
+            }
+            
+            let subscriptionKey = args["subscriptionKey"] as? String ?? ""
+            let region = args["region"] as? String ?? ""
+            let supportedLanguages = args["supportedLanguages"] as? [String] ?? ["zh-CN"]
+            let useExternalAudio = args["useExternalAudio"] as? Bool ?? false
+            
+            // 获取翻译服务配置参数
+            let translationAccessKey = args["translationAccessKey"] as? String ?? ""
+            let translationSecretKey = args["translationSecretKey"] as? String ?? ""
+            let translationRegion = args["translationRegion"] as? String ?? "cn-north-1"
+            
+            os_log("初始化AST服务", log: log, type: .info)
+            
+            // 创建Azure配置
+            let azureConfig = AzureConfiguration(
+                subscriptionKey: subscriptionKey,
+                region: region
+            )
+            
+            // 创建翻译配置
+            let translationConfig = TranslationConfiguration(
+                accessKey: translationAccessKey,
+                secretKey: translationSecretKey,
+                region: translationRegion
+            )
+            
+            // 创建服务配置
+            let serviceConfig = IntegratedSpeechTranslationService.ServiceConfiguration(
+                sourceLanguage: supportedLanguages.first ?? "zh-CN",
+                targetLanguage: supportedLanguages.count > 1 ? supportedLanguages[1] : "en-US"
+            )
+            
+            // 创建事件回调
+            let callback = AstEventCallback(plugin: self)
+            astEventCallback = callback
+            
+            // 使用异步任务调用初始化方法
+            Task {
+                await azureAstHelper.initialize(
+                    azureConfig: azureConfig,
+                    translationConfig: translationConfig,
+                    serviceConfig: serviceConfig,
+                    callback: callback
+                )
+            }
+            result(true)
+            
+        default:
+            result(FlutterMethodNotImplemented)
+        }
+    }
+    
 }
-
+// MARK: - AST Event Callback
-// MARK: - ASR回调包装器
+/**
-private class AsrCallbackWrapper: AzureAsrHelper.ContinuousRecognizeCallback {
+ * AST事件回调类
+ * 实现 IntegratedSpeechTranslationService.ServiceEventCallback 协议
+ */
+private class AstEventCallback: IntegratedSpeechTranslationService.ServiceEventCallback {
     private weak var plugin: AzureSpeechPlugin?
     
+    /**
+     * 构造函数
+     * @param plugin AzureSpeechPlugin 实例的弱引用
+     */
     init(plugin: AzureSpeechPlugin) {
         self.plugin = plugin
     }
     
-    func onResult(_ text: String, _ detectedLanguage: String) {
+    /**
-        plugin?.sendAsrEvent([
+     * 服务初始化完成回调
-            "type": "result",
+     */
-            "text": text,
+    public func onServiceInitialized() {
-            "detectedLanguage": detectedLanguage
+//        plugin?.sendAsrEvent([
-        ])
+//            "type": "serviceInitialized"
+//        ])
     }
     
-    func onRecognizing(_ recognizing: String, _ detectedLanguage: String) {
+    /**
-        plugin?.sendAsrEvent([
+     * 识别中回调
-            "type": "recognizing",
+     * @param text 正在识别的文本
-            "text": recognizing,
+     * @param language 语言
-            "detectedLanguage": detectedLanguage
+     * @param confidence 置信度
-        ])
+     */
+    public func onRecognizing(text: String, language: String, confidence: Float) {
+//        plugin?.sendAsrEvent([
+//            "type": "recognizing",
+//            "text": text,
+//            "language": language,
+//            "confidence": confidence
+//        ])
     }
     
-    func onSessionStarted() {
+    /**
-        plugin?.sendAsrEvent([
+     * 识别完成回调
-            "type": "sessionStarted"
+     * @param text 识别的文本
-        ])
+     * @param language 语言
+     * @param confidence 置信度
+     */
+    public func onRecognized(text: String, language: String, confidence: Float) {
+           print( "识别到文本: \(text), 语言: \(language), 置信度: \(confidence)")
+       plugin?.sendAsrEvent([
+           "type": "result1",
+           "text": text,
+           "language": language,
+       ])
     }
     
-    func onSessionStopped() {
+    /**
-        plugin?.sendAsrEvent([
+     * 翻译完成回调
-            "type": "sessionStopped"
+     * @param originalText 原始文本
-        ])
+     * @param translatedText 翻译文本
+     * @param targetLanguage 目标语言
+     */
+   public func onTranslated(originalText: String, translatedText: String, targetLanguage: String) {
+//        plugin?.sendAsrEvent([
+//            "type": "translated",
+//            "originalText": originalText,
+//            "translatedText": translatedText,
+//            "targetLanguage": targetLanguage
+//        ])
     }
     
-    func onCanceled(_ reason: String, _ errorDetails: String) {
+    /**
-        plugin?.sendAsrEvent([
+     * 翻译开始回调
-            "type": "canceled",
+     * @param text 要翻译的文本
-            "reason": reason,
+     */
-            "errorDetails": errorDetails
+    public func onTranslationStarted(text: String) {
-        ])
+//        plugin?.sendAsrEvent([
+//            "type": "translationStarted",
+//            "text": text
+//        ])
     }
     
-    func onError(_ error: String) {
+    /**
-        plugin?.sendAsrEvent([
+     * 翻译失败回调
-            "type": "error",
+     * @param text 翻译失败的文本
-            "error": error
+     * @param error 错误信息
-        ])
+     */
+   public func onTranslationFailed(text: String, error: String) {
+//        plugin?.sendAsrEvent([
+//            "type": "translationFailed",
+//            "text": text,
+//            "error": error
+//        ])
+    }
+    
+    /**
+     * 语音合成开始回调
+     * @param text 要合成的文本
+     */
+    func onSynthesisStarted(text: String) {
+//        plugin?.sendAsrEvent([
+//            "type": "synthesisStarted",
+//            "text": text
+//        ])
+    }
+    
+    /**
+     * 语音合成完成回调
+     * @param text 合成的文本
+     */
+   public func onSynthesisCompleted(text: String) {
+//        plugin?.sendAsrEvent([
+//            "type": "synthesisCompleted",
+//            "text": text
+//        ])
+    }
+    
+    /**
+     * 语音合成失败回调
+     * @param text 合成失败的文本
+     * @param error 错误信息
+     */
+   public func onSynthesisFailed(text: String, error: String) {
+//        plugin?.sendAsrEvent([
+//            "type": "synthesisFailed",
+//            "text": text,
+//            "error": error
+//        ])
+    }
+    
+    /**
+     * 语音合成进度回调
+     * @param text 正在合成的文本
+     * @param progress 进度（0.0-1.0）
+     */
+   public func onSynthesisProgress(text: String, progress: Float) {
+//        plugin?.sendAsrEvent([
+//            "type": "synthesisProgress",
+//            "text": text,
+//            "progress": progress
+//        ])
+    }
+    
+    /**
+     * 识别开始回调
+     */
+    func onRecognitionStarted() {
+//        plugin?.sendAsrEvent([
+//            "type": "recognitionStarted"
+//        ])
+    }
+    
+    /**
+     * 识别停止回调
+     */
+    public func onRecognitionStopped() {
+//        plugin?.sendAsrEvent([
+//            "type": "recognitionStopped"
+//        ])
+    }
+    
+    /**
+     * 状态变化回调
+     * @param component 组件名称
+     * @param isActive 是否激活
+     */
+    public func onStateChanged(component: String, isActive: Bool) {
+//        plugin?.sendAsrEvent([
+//            "type": "stateChanged",
+//            "component": component,
+//            "isActive": isActive
+//        ])
+    }
+    
+    /**
+     * 错误回调
+     * @param component 组件名称
+     * @param error 错误信息
+     */
+    public func onError(component: String, error: String) {
+//        plugin?.sendAsrEvent([
+//            "type": "error",
+//            "component": component,
+//            "error": error
+//        ])
+    }
+    
+    /**
+     * 语音合成音频数据生成回调
+     * @param text 合成的文本
+     * @param audioData 音频数据
+     */
+    public func onSynthesisAudioGenerated(text: String, audioData: Data) {
+        print("语音合成音频数据生成回调", text, audioData.count)
+        // 调用BLE服务写入音频数据
+        BleService.shared.writeExternalAudioData(data: audioData)
+    
+//        plugin?.sendAsrEvent([
+//            "type": "synthesisAudioGenerated",
+//            "text": text,
+//            "audioDataSize": audioData.count
+//        ])
     }
 }
 
+extension AzureSpeechPlugin: BleService.Callback {
+    /// 扫描结果回调
+    /// - Parameter devices: 扫描到的设备列表
+    public func onScanResult(devices: [[String: Any]]) {
+    }
+    
+    /// 连接状态变化回调
+    /// - Parameter state: 连接状态
+    public func onConnectionStateChanged(state: Int) {
+    }
+    
+    /// 接收到音频数据回调
+    /// - Parameter data: 音频数据
+    public func onAudioDataReceived(data: Data) {
+       // 安全解包版本
+            guard let audioStream = azureAsrHelper.audioStream else {
+                os_log("音频流未初始化", type: .error)
+                return
+            }
+            audioStream.saveAudioDataTo(data: data)
+    }
+    
+    /// 接收到音频数据回调1
+    /// - Parameter data: 音频数据
+    public func onAudioDataReceived1(data: Data) {
+       // 安全解包版本
+            guard let audioStream = azureAstHelper.audioProcessor else {
+                os_log("音频流未初始化", type: .error)
+                return
+            }
+            audioStream.saveAudioDataTo(data: data)
+    }
+    
+    /// 接收到编码数据回调
+    /// - Parameter data: 编码数据
+    public func onEncodedDataReceived(data: Data) {
+
+    }
+    
+    /// 唤醒信号接收回调
+    public func onWakeupSignalReceived() {
+      
+    }
+    
+    /// 设备信息接收回调
+    /// - Parameters:
+    ///   - infoType: 信息类型
+    ///   - infoData: 信息数据
+    public func onDeviceInfoReceived(infoType: Int, infoData: [String: Any]) {
+    }
+}
 // MARK: - 事件处理
 extension AzureSpeechPlugin: FlutterStreamHandler {
     public func onListen(withArguments arguments: Any?, eventSink events: @escaping FlutterEventSink) -> FlutterError? {
@@ -596,6 +956,7 @@ extension AzureSpeechPlugin: AudioDataListener {
     }
 }
 
+
 // MARK: - 音频数据回调实现
 /**
  * 音频数据回调实现类
@@ -621,4 +982,89 @@ private class AudioDataCallbackImpl: SimpleAudioReceiver.AudioDataCallback {
     }
 }
 
+// MARK: - ASR Callback Wrapper
+/**
+ * ASR回调包装器类
+ * 实现 AzureAsrHelper.ContinuousRecognizeCallback 协议
+ */
+private class AsrCallbackWrapper: AzureAsrHelper.ContinuousRecognizeCallback {
+    private weak var plugin: AzureSpeechPlugin?
+    
+    /**
+     * 初始化ASR回调包装器
+     * @param plugin AzureSpeechPlugin实例
+     */
+    init(plugin: AzureSpeechPlugin) {
+        self.plugin = plugin
+    }
+    
+    /**
+     * 识别结果回调
+     * @param text 识别的文本
+     * @param detectedLanguage 检测到的语言
+     */
+    func onResult(_ text: String, _ detectedLanguage: String) {
+        plugin?.sendAsrEvent([
+            "type": "result",
+            "text": text,
+            "detectedLanguage": detectedLanguage
+        ])
+    }
+    
+    /**
+     * 识别中回调
+     * @param text 正在识别的文本
+     * @param detectedLanguage 检测到的语言
+     */
+    func onRecognizing(_ text: String, _ detectedLanguage: String) {
+        plugin?.sendAsrEvent([
+            "type": "recognizing",
+            "text": text,
+            "detectedLanguage": detectedLanguage
+        ])
+    }
+    
+    /**
+     * 会话开始回调
+     */
+    func onSessionStarted() {
+        plugin?.sendAsrEvent([
+            "type": "sessionStarted"
+        ])
+    }
+    
+    /**
+     * 会话结束回调
+     */
+    func onSessionStopped() {
+        plugin?.sendAsrEvent([
+            "type": "sessionStopped"
+        ])
+    }
+    
+    /**
+     * 识别取消回调
+     * @param reason 取消原因
+     * @param errorDetails 错误详情
+     */
+    func onCanceled(_ reason: String, _ errorDetails: String) {
+        plugin?.sendAsrEvent([
+            "type": "canceled",
+            "reason": reason,
+            "errorDetails": errorDetails
+        ])
+    }
+    
+    /**
+     * 错误回调
+     * @param error 错误信息
+     */
+    func onError(_ error: String) {
+        plugin?.sendAsrEvent([
+            "type": "error",
+            "error": error
+        ])
+    }
+}
+
 

local_plugins/azure_speech/ios/azure_speech/Sources/azure_speech/IntegratedSpeechTranslationService.swift

@@ -44,7 +44,7 @@ import os.log
     private var translationService: TranslationServiceInterface?
     
     // 音频处理
-    private var audioProcessor: SimpleAudioReceiver?
+    public var audioProcessor: SimpleAudioReceiver?
     private var audioConfig: SPXAudioConfiguration?
     
     // 录音文件处理
@@ -333,7 +333,7 @@ import os.log
      * 初始化音频处理器
      */
     private func initializeAudioProcessor() {
-        audioProcessor = SimpleAudioReceiver(parentHelper: self as! AzureAsrHelper)
+        audioProcessor = SimpleAudioReceiver()
         audioProcessor?.initAudioRecord()
         // 检查音频配置是否已存在
         if audioConfig == nil, let pushStream = audioProcessor?.pushAudioStream {
@@ -486,16 +486,24 @@ import os.log
                 }
             }
             
-            // 合成进行中事件
+            // 合成进行中事件 - 修改为返回音频数据
             synthesizer?.addSynthesizingEventHandler { [weak self] (synthesizer, event) in
                 guard let self = self else { return }
                 
-                os_log("合成进行中事件", log: self.log, type: .debug)
+                // 安全解包audioData
-                // 计算进度（简化版）
+                guard let audioData = event.result.audioData else {
-                let progress: Float = 0.5
+                    os_log("合成进行中事件，音频数据为空", log: self.log, type: .debug)
+                    return
+                }
+                
+                os_log("合成进行中事件，音频数据长度: %d bytes", log: self.log, type: .debug, audioData.count)
+                
+              
                 
                 DispatchQueue.main.async {
-                    self.eventCallback?.onSynthesisProgress(text: "", progress: progress)
+             
+                    // 立即返回当前生成的音频数据
+                    self.eventCallback?.onSynthesisAudioGenerated(text: "", audioData: audioData)
                 }
             }
             
@@ -506,6 +514,7 @@ import os.log
                 os_log("合成完成事件", log: self.log, type: .info)
                 
                 DispatchQueue.main.async {
+                    self.serviceState.isSynthesizing = false
                     self.eventCallback?.onStateChanged(component: "Synthesis", isActive: false)
                 }
             }
@@ -654,10 +663,8 @@ import os.log
                 }
                 
                 // 进行语音合成
-                if let audioData = synthesizeText(translatedText) {
+                if synthesizeTextStreaming(translatedText) {
-                    await MainActor.run {
+                    print("语音合成已启动")
-                        eventCallback?.onSynthesisAudioGenerated(text: translatedText, audioData: audioData)
-                    }
                 }
             } else {
                 let errorMessage = translationResult?.error ?? "翻译失败"
@@ -688,54 +695,34 @@ import os.log
     /**
      * 合成文本为音频数据
      */
-    private func synthesizeText(_ text: String) -> Data? {
+    /// 流式语音合成
+    /// - Parameter text: 要合成的文本
+    /// - Returns: 合成是否成功启动
+    private func synthesizeTextStreaming(_ text: String) -> Bool {
         guard let synthesizer = synthesizer else {
             os_log("语音合成器未初始化", log: log, type: .error)
-            return nil
+            return false
         }
         
         do {
             let ssml = buildSSML(text: text)
-            os_log("开始合成语音: %@", log: log, type: .info, text)
+            os_log("开始流式合成语音: %@", log: log, type: .info, text)
-            
+          
             serviceState.isSynthesizing = true
             
             DispatchQueue.main.async {
                 self.eventCallback?.onSynthesisStarted(text: text)
                 self.eventCallback?.onStateChanged(component: "Synthesis", isActive: true)
             }
+            try synthesizer.speakSsml(ssml)
+           
             
-            let result = try synthesizer.speakSsml(ssml)
+            return true
-            
-            serviceState.isSynthesizing = false
-            
-            DispatchQueue.main.async {
-                self.eventCallback?.onStateChanged(component: "Synthesis", isActive: false)
-            }
-            
-            if result.reason == .synthesizingAudioCompleted {
-                os_log("语音合成成功", log: log, type: .info)
-                
-                DispatchQueue.main.async {
-                    self.eventCallback?.onSynthesisCompleted(text: text)
-                }
-                
-                return result.audioData
-            } else {
-                let errorMessage = "合成失败: \(result.reason)"
-                os_log("%@", log: log, type: .error, errorMessage)
-                
-                DispatchQueue.main.async {
-                    self.eventCallback?.onSynthesisFailed(text: text, error: errorMessage)
-                }
-                
-                return nil
-            }
             
         } catch {
             serviceState.isSynthesizing = false
             
-            let errorMessage = "合成异常: \(error.localizedDescription)"
+            let errorMessage = "合成启动异常: \(error.localizedDescription)"
             os_log("%@", log: log, type: .error, errorMessage)
             
             DispatchQueue.main.async {
@@ -743,10 +730,17 @@ import os.log
                 self.eventCallback?.onStateChanged(component: "Synthesis", isActive: false)
             }
             
-            return nil
+            return false
         }
     }
     
+    /// 保留原有的同步合成方法作为备用
+    private func synthesizeText(_ text: String) -> Data? {
+    // 现在调用流式合成方法
+    let success = synthesizeTextStreaming(text)
+    return success ? Data() : nil // 返回空数据表示已启动，实际数据通过回调返回
+    }
+    
     /**
      * 构建SSML
      */

local_plugins/azure_speech/ios/azure_speech/Sources/tools/SimpleAudioReceiver.swift

@@ -38,14 +38,8 @@ public class SimpleAudioReceiver: NSObject {
     private var currentRoute: AudioOutputRoute?
     //public var onAudioData: ((Data) -> Void)?
     public var recordfile: RecordFile?
-    // 添加对外部类的弱引用
+
-    private weak var parentHelper: AzureAsrHelper?
+
-    
-    // 添加初始化方法，接收外部类引用
-    init(parentHelper: AzureAsrHelper) {
-        self.parentHelper = parentHelper
-        super.init()
-    }
     /**
      * 初始化
      */
@@ -170,7 +164,7 @@ public class SimpleAudioReceiver: NSObject {
      */
     public func saveAudioDataTo(data: Data) {
         // 通过父类引用调用方法
-        if audioSourceType != .external || !(parentHelper?.isContinuousRecognitionActive() ?? false) {
+        if audioSourceType != .external {
             return
         }
         // print("外部data=\(data)")

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/Info.plist

@@ -0,0 +1,27 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>AvailableLibraries</key>
+	<array>
+		<dict>
+			<key>BinaryPath</key>
+			<string>JLAudioUnitKit.framework/JLAudioUnitKit</string>
+			<key>LibraryIdentifier</key>
+			<string>ios-arm64</string>
+			<key>LibraryPath</key>
+			<string>JLAudioUnitKit.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>arm64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+		</dict>
+	</array>
+	<key>CFBundlePackageType</key>
+	<string>XFWK</string>
+	<key>XCFrameworkFormatVersion</key>
+	<string>1.0</string>
+</dict>
+</plist>

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLAudioUnitKit.h

@@ -0,0 +1,24 @@
+//
+//  JLAudioUnitKit.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2024/11/14.
+//
+
+#import <Foundation/Foundation.h>
+
+//! Project version number for JLAudioUnitKit.
+FOUNDATION_EXPORT double JLAudioUnitKitVersionNumber;
+
+//! Project version string for JLAudioUnitKit.
+FOUNDATION_EXPORT const unsigned char JLAudioUnitKitVersionString[];
+
+// In this header, you should import all the public headers of your framework using statements like #import <JLAudioUnitKit/PublicHeader.h>
+
+#import <JLAudioUnitKit/JLSpeexDecoder.h>
+#import <JLAudioUnitKit/JLOpusDecoder.h>
+#import <JLAudioUnitKit/JLOpusEncoder.h>
+#import <JLAudioUnitKit/JLOpusFormat.h>
+#import <JLAudioUnitKit/JLPcmToWtg.h>
+#import <JLAudioUnitKit/JLPcmToWav.h>
+#import <JLAudioUnitKit/JLAudioUnitPlayer.h>

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLAudioUnitPlayer.h

@@ -0,0 +1,58 @@
+//
+//  JLAudioUnitPlayer.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2025/4/16.
+//  Copyright © 2025 ZhuHai JieLi Technology Co.,Ltd. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+#import <AudioToolbox/AudioToolbox.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSInteger, JLAudioUnitPlayerType) {
+    JLAudioUnitPlayerTypeFile,   // 支持 MP3/WAV/AAC 等文件格式（AVAudioPlayer）
+    JLAudioUnitPlayerTypePCM      // 支持 PCM 流式播放（Audio Queue）
+};
+
+@class JLAudioUnitPlayer;
+
+@protocol JLAudioPlayerDelegate <NSObject>
+@optional
+// 播放进度更新（秒）
+- (void)audioPlayer:(JLAudioUnitPlayer *)player didUpdateProgress:(NSTimeInterval)currentTime duration:(NSTimeInterval)duration;
+// 播放完成
+- (void)audioPlayerDidFinishPlaying:(JLAudioUnitPlayer *)player;
+// 播放错误
+- (void)audioPlayer:(JLAudioUnitPlayer *)player didFailWithError:(NSError *)error;
+@end
+
+@interface JLAudioUnitPlayer : NSObject
+
+@property (nonatomic, weak) id<JLAudioPlayerDelegate> delegate;
+@property (nonatomic, readonly) NSTimeInterval duration;  // 总时长（文件模式下有效）
+@property (nonatomic, readonly) NSTimeInterval currentTime;
+@property (nonatomic, readonly) BOOL isPlaying;
+
+#pragma mark - 初始化方法
+
+/// 文件播放初始化（MP3/WAV/AAC）
+- (instancetype)initWithAudioFile:(NSString *)filePath;
+
+/// PCM 流播放初始化（需指定格式）
+- (instancetype)initWithPCMFormat:(AudioStreamBasicDescription)format;
+
+#pragma mark - 播放控制
+- (void)play;
+- (void)pause;
+- (void)stop;
+- (void)seekToTime:(NSTimeInterval)time; // 仅文件模式有效
+
+#pragma mark - PCM 流式输入（仅 PCM 模式有效）
+- (void)appendPCMData:(NSData *)pcmData;
+- (void)endPCMStream; // 结束 PCM 流输入
+
+@end
+
+NS_ASSUME_NONNULL_END

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLOpusDecoder.h

@@ -0,0 +1,66 @@
+//
+//  JLOpusDecoder.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2024/11/14.
+//  Copyright © 2024 ZhuHai JieLi Technology Co.,Ltd. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+#import <JLAudioUnitKit/JLOpusFormat.h>
+
+NS_ASSUME_NONNULL_BEGIN
+@class JLOpusDecoder;
+
+/// Opus 解码代理
+@protocol JLOpusDecoderDelegate <NSObject>
+
+/// Opus 数据解码
+/// - Parameters:
+///   - decoder: 解码器
+///   - data: pcm 数据
+///   - error: 错误信息
+-(void)opusDecoder:(JLOpusDecoder *)decoder Data:(NSData* _Nullable)data error:(NSError* _Nullable)error;
+
+@end
+
+typedef void(^JLOpusDecoderConvertBlock)(NSString *_Nullable pcmPath,NSError *_Nullable error);
+
+/// Opus 解码
+@interface JLOpusDecoder : NSObject
+
+/// 数据格式参数
+@property (nonatomic, strong) JLOpusFormat *opusFormat;
+
+/// 代理委托
+@property(nonatomic, weak) id<JLOpusDecoderDelegate> delegate;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/// 初始化
+/// - Parameters:
+///   - format: 解码格式
+///   - delegate: 代理
+- (instancetype)initDecoder:(JLOpusFormat *)format delegate:(id<JLOpusDecoderDelegate>)delegate;
+
+/// 重置解码格式
+/// - Parameter format: 解码格式
+-(void)resetOpusFramet:(JLOpusFormat *)format;
+
+/// 输入 Opus 数据
+/// - Parameter data: Opus 数据
+-(void)opusDecoderInputData:(NSData *)data;
+
+/// 解码文件
+/// - Parameters:
+///   - input: opus 文件
+///   - outPut: 输出路径
+///   - result: 结果回调
+-(void)opusDecodeFile:(NSString *)input outPut:(NSString *_Nullable)outPut Resoult:(JLOpusDecoderConvertBlock _Nullable)result;
+
+/// 释放
+-(void)opusOnRelease;
+
+@end
+
+NS_ASSUME_NONNULL_END

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLOpusEncoder.h

@@ -0,0 +1,62 @@
+//
+//  JLOpusEncoder.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2024/11/14.
+//  Copyright © 2024 ZhuHai JieLi Technology Co.,Ltd. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+#import <JLAudioUnitKit/JLOpusFormat.h>
+
+NS_ASSUME_NONNULL_BEGIN
+@class JLOpusEncoder;
+
+/// Opus 编码代理
+@protocol JLOpusEncoderDelegate <NSObject>
+
+/// PCM 数据编码
+/// - Parameters:
+///   - encoder: 解码器
+///   - data: opus 数据
+///   - error: 错误信息
+-(void)opusEncoder:(JLOpusEncoder *)encoder Data:(NSData* _Nullable)data error:(NSError* _Nullable)error;
+
+@end
+
+typedef void(^JLOpusEncoderConvertBlock)(NSString *_Nullable pcmPath,NSError *_Nullable error);
+
+/// Opus 编码
+@interface JLOpusEncoder : NSObject
+
+/// 音频格式
+@property (nonatomic, strong) JLOpusFormat *opusFormat;
+
+/// 代理
+@property (nonatomic, weak) id<JLOpusEncoderDelegate> delegate;
+
+-(instancetype)init NS_UNAVAILABLE;
+
+/// 初始化
+/// - Parameters:
+///   - format: 音频格式
+///   - delegate: 代理
+-(instancetype)initFormat:(JLOpusFormat *)format delegate:(id<JLOpusEncoderDelegate>)delegate;
+
+/// PCM 数据
+/// - Parameter data: PCM 数据
+-(void)opusEncodeData:(NSData *)data;
+
+/// PCM 文件转换成 Opus 文件
+/// - Parameters:
+///   - pcmPath: PCM 文件存放路径
+///   - outPut: 文件输出路径
+///   - result: 结果回调
+-(void)opusEncodeFile:(NSString *)pcmPath outPut:(NSString *_Nullable)outPut Resoult:(JLOpusEncoderConvertBlock _Nullable)result;
+
+/// 释放
+-(void)opusOnRelease;
+
+@end
+
+NS_ASSUME_NONNULL_END

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLOpusFormat.h

@@ -0,0 +1,53 @@
+//
+//  JLOpusOptions.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2024/11/14.
+//  Copyright © 2024 ZhuHai JieLi Technology Co.,Ltd. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+#define OPUS_JL_MAX_FRAME_SIZE 48000 * 2
+#define OPUS_JL_MAX_PACKET_SIZE 1500
+
+
+@interface JLOpusFormat : NSObject
+
+/// 采样率
+@property (nonatomic, assign) int sampleRate;
+
+/// 单/双声道
+@property (nonatomic, assign) int channels;
+
+/// 帧长度 20ms 默认值
+@property (nonatomic, assign) int frameDuration;
+
+/// BitRate
+@property (nonatomic, assign) int bitRate;
+
+/// 数据帧大小
+@property (nonatomic, assign, readonly) int frameSize;
+
+/// 数据大小
+@property (nonatomic, assign) int dataSize;
+
+/// 是否包含数据头部
+@property (nonatomic, assign) BOOL hasDataHeader;
+
+-(instancetype)init NS_UNAVAILABLE;
+
+/// 默认配置
+/// sampleRate: 16000
+/// channels: 1
+/// frameDuration: 20
+/// dataSize: 40
+/// frameSize: 320
+/// hasDataHeader: YES
++(JLOpusFormat*)defaultFormats;
+
+@end
+
+NS_ASSUME_NONNULL_END

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLPcmToWav.h

@@ -0,0 +1,55 @@
+//
+//  JLPcmToWav.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2025/4/15.
+//  Copyright © 2025 ZhuHai JieLi Technology Co.,Ltd. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/// PCM 转 WAV
+@interface JLPcmToWav : NSObject
+
+#pragma mark - 流式编码接口
+
+/// 初始化流式编码器
+/// @param outputPath 输出文件路径
+/// @param sampleRate 采样率（如 16000）
+/// @param numChannels 声道数（1 或 2）
+/// @param bitsPerSample 位深（16）
+- (instancetype)initWithOutputPath:(NSString *)outputPath
+                       sampleRate:(uint32_t)sampleRate
+                     numChannels:(uint16_t)numChannels
+                  bitsPerSample:(uint16_t)bitsPerSample;
+
+/// 追加 PCM 数据（需确保数据格式与初始化参数一致）
+/// @param pcmData PCM 数据块
+/// @param error 错误信息
+- (BOOL)appendPCMData:(NSData *)pcmData error:(NSError **)error;
+
+/// 完成编码并关闭文件（必须调用以更新 WAV 头）
+/// @param error 错误信息
+- (BOOL)finishWithError:(NSError **)error;
+
+#pragma mark - 一次性编码接口（便捷方法）
+
+/// 将 PCM 数据一次性转为 WAV 文件
+/// @param pcmData PCM 数据块
+/// @param outputPath 输出文件路径
+/// @param sampleRate 采样率（如 16000）
+/// @param numChannels 声道数（1 或 2）
+/// @param bitsPerSample 位深（16）
+/// @param error 错误信息
++ (BOOL)convertPCMData:(NSData *)pcmData
+            toWAVFile:(NSString *)outputPath
+           sampleRate:(uint32_t)sampleRate
+         numChannels:(uint16_t)numChannels
+      bitsPerSample:(uint16_t)bitsPerSample
+              error:(NSError **)error;
+
+@end
+
+NS_ASSUME_NONNULL_END

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLPcmToWtg.h

@@ -0,0 +1,51 @@
+//
+//  JLPcmToWtg.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2025/1/21.
+//  Copyright © 2025 ZhuHai JieLi Technology Co.,Ltd. All rights reserved.
+//
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface JLPcm2WtgModel : NSObject <NSCopying>
+
+/// 音频文件
+@property (nonatomic, copy) NSString *pcmPath;
+
+/// 输出文件
+@property (nonatomic, copy) NSString *wtgPath;
+
+@end
+
+
+/// JLPcmToWtgProtocol
+@protocol JLPcmToWtgDelegate <NSObject>
+
+/// 音频转码完成
+/// - Parameter model: 音频转码模型
+- (void)convertPcmToWtgDone:(JLPcm2WtgModel *)model;
+
+@end
+
+/// 音频转码
+/// PCM 转 WTG
+@interface JLPcmToWtg : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/// 音频转码初始化
+/// - Parameter delegate: 音频转码代理
+- (instancetype)initWithDelegate:(id<JLPcmToWtgDelegate>)delegate;
+
+/// 音频转码
+/// 当前接口模型只限制 PCM 文件格式为：8k 16bit
+/// 输入的pcm文件（16bit,小端，8k采样率），wtg_filename：输出的wtg文件
+/// - Parameter model: 音频转码模型
+-(void)convertPcmToWtg:(JLPcm2WtgModel *)model;
+
+@end
+
+NS_ASSUME_NONNULL_END

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Headers/JLSpeexDecoder.h

@@ -0,0 +1,54 @@
+//
+//  JLSpeexUnit.h
+//  JLAudioUnitKit
+//
+//  Created by EzioChan on 2024/11/14.
+//
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+@class JLSpeexDecoder;
+
+@protocol JLSpeexDelegate <NSObject>
+
+/// Speex 数据解码
+/// - Parameters:
+///   - decoder: 解码器
+///   - data: PCM 数据
+///   - error: 错误信息
+- (void)speexDecoder:(JLSpeexDecoder *)decoder Data:(NSData* _Nullable)data error:(NSError* _Nullable)error;
+
+@end
+
+typedef void(^JLSpeexConvertBlock)(NSString *_Nullable pcmPath,NSError *_Nullable error);
+
+/// Speex 数据解码
+@interface JLSpeexDecoder : NSObject
+
+-(instancetype)init NS_UNAVAILABLE;
+
+/// 初始化
+/// - Parameter delegate: 委托
+-(instancetype)initWithDelegate:(id<JLSpeexDelegate>)delegate;
+
+/// 委托
+@property (nonatomic, weak) id<JLSpeexDelegate> delegate;
+
+/// 输入 Speex 数据
+/// - Parameter data: Speex 数据
+-(void)speexInputData:(NSData*)data;
+
+/// 将 Speex 转换为 PCM
+/// - Parameters:
+///   - filePath: Speex 路径
+///   - opPath: 输出路径
+///   - result: 结果回调
+-(void)speexConvertToPcm:(NSString *)filePath outPutFilePath:(NSString * _Nullable)opPath Result:(JLSpeexConvertBlock)result;
+
+/// 释放对象
+-(void)speexOnRelease;
+
+@end
+
+NS_ASSUME_NONNULL_END

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/Modules/module.modulemap

@@ -0,0 +1,6 @@
+framework module JLAudioUnitKit {
+  umbrella header "JLAudioUnitKit.h"
+  export *
+
+  module * { export * }
+}

local_plugins/ble_service/ios/ble_service/JLAudioUnitKit.xcframework/ios-arm64/JLAudioUnitKit.framework/readme.txt

@@ -0,0 +1,52 @@
+DESCRIPTION
+-----------
+This package includes the files needed to build the fixed point implementation
+of the ITU-T G.729 Appendix IV.
+
+DIFFERENCES WITH G.729B
+-----------------------
+The ANSI-C source code simulating the bit-exact, fixed point simulation 
+software of enhanced VAD is integrated into the G.729B source code. 
+The modification to the G.729B source code updated files of coder.c, 
+cod_ld8k.c, bits.c, dec_sid.c, dtx.c, qsidgain.c, qsidlsf.c, tab_dtx.c, 
+tab_dtx.h, ld8k.h, deleted vad.c/h and integrated the enhanced VAD functions
+into the G.729B project. The modification also replaced the original basic 
+operators used in the original G.729B source code by the basic operators of 
+ITU-T Software Tool Library STL2005.
+
+COMPILATION
+-----------
+For UNIX systems the following makefiles are provided
+
+   coder.mak
+   decoder.mak
+
+Edit the makefiles coder.mak and decoder.mak to set the proper options
+for your system.
+The command to compile and link all code on a UNIX system is
+
+   make -f coder.mak
+   make -f decoder.mak
+
+For other platforms, the *.mak files can be used to work out the
+compilation procedures.
+
+USAGE
+-----
+The command line instruction for the encoder is as follows:
+   coder inputfile bitstreamfile dtx_option mode x 
+where:
+   inputfile    : 8 kHz sampled data file containing 16 bit linear PCM signal 
+   bitstreamfile: binary file containing bitstream 
+   dtx_option   : = 1 : DTX enabled   0 : DTX disabled
+   mode X       : Run encoder with enhanced VAD in operating point X, 
+                  X=0 or 1 or 2 representing balanced or quality-preferred 
+                  or bandwidth saving operating point, respectively  
+
+The command line for the decoder is as follows:
+   decoder bitstreamfile outputfile
+where:
+   bitstreamfile: binary file containing bitstream
+   outputfile   : 8 kHz sampled data file containing 16 bit linear PCM signal
+
+

local_plugins/ble_service/ios/ble_service/Package.swift

@@ -8,22 +8,34 @@ let package = Package(
     .library(name: "ble-service", targets: ["ble_service"])
   ],
   dependencies: [
-    // YbridOpus SPM 包（替代CocoaPods依赖）
-    .package(url: "https://github.com/ybrid/opus-swift.git", from: "0.8.0")
   ],
   targets: [
     .target(
       name: "ble_service",
       dependencies: [
-        .product(name: "YbridOpus", package: "opus-swift")
+        "opus",
+        "SpeexKit",
+        "JLAudioUnitKit"
       ],
       path: "Sources/ble_service",
       linkerSettings: [
         .linkedFramework("AVFoundation"),
         .linkedFramework("AudioToolbox"),
         .linkedFramework("CoreBluetooth")
-        // 移除 AccessorySetupKit 链接，因为它只在 iOS 18+ 可用
       ]
+    ),
+    // 本地opus框架作为二进制目标（使用xcframework格式）
+    .binaryTarget(
+      name: "opus", // 改为与 xcframework 内部模块名称一致
+      path: "opus.xcframework"
+    ),
+    .binaryTarget(
+      name: "SpeexKit",
+      path: "SpeexKit.xcframework"
+    ),  // 添加了缺失的逗号
+    .binaryTarget(
+      name: "JLAudioUnitKit",
+      path: "JLAudioUnitKit.xcframework"
     )
   ]
 )

local_plugins/ble_service/ios/ble_service/Sources/ble_service/BleConst.swift

@@ -20,7 +20,12 @@ class BleConst {
     
     /** 通知特征UUID - 文档中定义为0000ABC2-0000-1111-2222-123456789ABC */
     static let NOTIFY_CHAR_UUID = CBUUID(string: "0000abc2-0000-1111-2222-123456789abc")
-    
+    /** 通话音频服务UUID - 文档中定义为0000ABC0-0001-1111-2222-123456789ABC */
+
+    static let CALL_AUDIO_SERVICE_UUID = CBUUID(string: "0000abc0-0001-1111-2222-123456789abc")
+
+    /** 通话写入音频特征UUID - 文档中定义为0000ABC1-0001-1111-2222-123456789ABC */
+    static let CALL_WRITE_AUDIO_CHAR_UUID = CBUUID(string: "0000abc1-0001-1111-2222-123456789abc")
     /** 客户端特征配置描述符UUID */
     static let CLIENT_CHAR_CONFIG_UUID = CBUUID(string: "00002902-0000-1000-8000-00805f9b34fb")
     

local_plugins/ble_service/ios/ble_service/Sources/ble_service/BleService.swift

@@ -28,6 +28,7 @@ public class BleService: NSObject {
         
         // 数据相关回调
         func onAudioDataReceived(data: Data)
+        func onAudioDataReceived1(data: Data)
         
         // 唤醒信号回调
         func onWakeupSignalReceived()
@@ -40,9 +41,14 @@ public class BleService: NSObject {
     private var centralManager: CBCentralManager?
     private var peripheralDevice: CBPeripheral?
     private var writeCharacteristic: CBCharacteristic?
+    private var callAudioCharacteristic: CBCharacteristic?
+
     private var notifyCharacteristic: CBCharacteristic?
     private var audioCharacteristic: CBCharacteristic?
     
+    // 添加录制文件管理器
+    private var recordingFile: RecordingFile?
+    
     private var isScanning = false
     private var scanTimer: Timer?
     
@@ -80,6 +86,24 @@ public class BleService: NSObject {
         // 是否回复指令码
 private var cmdReplyType: UInt8 = 0
 
+    // 新增：音频配置参数
+    private var sampleRate: Int32 = 16000
+    private var channels: Int32 = 1
+    private var frameSize: Int32 = 640
+    
+    // MARK: - 音频数据发送相关属性
+    private var audioBuffer = [UInt8]() // 音频数据缓冲区
+    private var audioSendQueue = DispatchQueue(label: "com.yunqiinnovation.ble_service.audio_send", qos: .userInitiated)
+    private var audioDataQueue = [Data]() // 音频数据发送队列
+    private var isAudioSending = false // 音频发送状态
+    private var audioSendTimer: Timer? // 音频发送定时器
+    
+    // 音频发送配置常量
+    private let AUDIO_CHUNK_SIZE = 120 // 每次发送120字节
+    private let AUDIO_SEND_INTERVAL: TimeInterval = 0.06 // 60毫秒间隔
+    
+    private let audioQueueLock = NSLock() // 队列操作锁
+   
     //private var currentCommand: Command?
       // 命令结构体
     private struct Command {
@@ -121,16 +145,29 @@ private var cmdReplyType: UInt8 = 0
     }
     
     // 启动Opus解码流
-    private func startOpusDecoding() {
+    private func startOpusDecoding(sampleRate: Int32 = 16000, channels: Int32 = 1, frameSize: Int32 = 640) {
+        // 检查参数是否与预设值匹配，如果不匹配则停止当前解码
+        if sampleRate != self.sampleRate || channels != self.channels || frameSize != self.frameSize {
+            opusProcessor?.stopDecoding()
+        }
+
         let opusStarted = opusProcessor?.startDecoding(
-            sampleRate: 16000, // 设备端Opus编码采样率
+            sampleRate: sampleRate, // 设备端Opus编码采样率
-            channels: 1,      // 单声道
+            channels: channels,      // 单声道
-            frameSize: 640     // 40ms at 16kHz = 640 samples
+            frameSize: frameSize     // 40ms at 16kHz = 640 samples
         ) ?? false
-        
+       
         if !opusStarted {
             os_log("Opus解码流启动失败", log: logger, type: .error)
         }
+        else{
+            os_log("Opus解码流启动成功", log: logger, type: .info)
+             // 保存音频配置参数
+        self.sampleRate = sampleRate
+        self.channels = channels
+        self.frameSize = frameSize
+        
+        }
     }
     
     // MARK: - 回调管理
@@ -345,17 +382,18 @@ private var cmdReplyType: UInt8 = 0
         peripheralDevice?.delegate = self
         updateConnectionState(BleConst.STATE_CONNECTED)
         
-        if opusProcessor?.isDecoding() != true {
+       // if opusProcessor?.isDecoding() != true {
             startOpusDecoding()
-        }
+       // }
         
         // 清空特征引用，强制重新发现
         writeCharacteristic = nil
+        callAudioCharacteristic = nil
         notifyCharacteristic = nil
         audioCharacteristic = nil
         
         os_log("🔍 恢复会话时重新发现服务和特征", log: logger, type: .info)
-        peripheral.discoverServices([BleConst.PRIMARY_SERVICE_UUID, BleConst.AUDIO_SERVICE_UUID])
+        peripheral.discoverServices([BleConst.PRIMARY_SERVICE_UUID, BleConst.AUDIO_SERVICE_UUID, BleConst.CALL_AUDIO_SERVICE_UUID])
         
         os_log("✅ 已恢复设备会话: %@", log: logger, type: .info, peripheral.identifier.uuidString)
     }
@@ -369,6 +407,16 @@ private var cmdReplyType: UInt8 = 0
         
         if centralManager != nil {
             isManualDisconnect = true  // 标记为主动断开
+            
+            // 停止音频发送定时器
+            stopAudioSendTimer()
+            
+            // 清空音频缓冲区和队列
+            audioQueueLock.lock()
+            audioBuffer.removeAll()
+            audioDataQueue.removeAll()
+            audioQueueLock.unlock()
+            
             updateConnectionState(BleConst.STATE_DISCONNECTING)
             centralManager?.cancelPeripheralConnection(peripheral)
             os_log("🔌 开始断开设备连接", log: logger, type: .info)
@@ -481,6 +529,12 @@ private var cmdReplyType: UInt8 = 0
             callback.onAudioDataReceived(data: data)
         }
     }
+       /// 通知所有回调接收到音频数据
+    private func notifyAudioDataReceived1(data: Data) {
+        for callback in callbacks {
+            callback.onAudioDataReceived1(data: data)
+        }
+    }
     
     /// 通知所有回调接收到唤醒信号
     private func notifyWakeupSignalReceived() {
@@ -527,11 +581,23 @@ private var cmdReplyType: UInt8 = 0
     }
     
     // MARK: - 编解码控制
-    
+
+    public func writeExternalAudioData(data: Data) {
+        print("pcm音频进入编码", data.count)
+        // 确保Opus编码器处于活跃状态（如果不是则启动）
+     
+            // 将外部音频数据写入编码流，每次处理1280字节
+            opusProcessor?.encodePCMData(data)
+       
+    }
+
     /// 打开编码器
     func openEncoder() -> Bool {
         os_log("🎙️ 准备打开编码（设备侧）...", log: logger, type: .info)
-        
+          // 确保Opus解码器处于活跃状态（如果不是则启动）
+      //  if opusProcessor?.isDecoding() != true {
+            startOpusDecoding()
+        //}
         let paramData = Data([BleConst.CODEC_CONTROL_ENCODE_ON, BleConst.AUDIO_CHANNEL_RIGHT])
         let result = sendCommand(BleConst.CMD_CONTROL_CODEC, data: paramData)
         
@@ -549,25 +615,50 @@ private var cmdReplyType: UInt8 = 0
         os_log("打开解码...", log: logger, type: .info)
         
         // 确保Opus解码器处于活跃状态（如果不是则启动）
-        if opusProcessor?.isDecoding() != true {
+        //if opusProcessor?.isDecoding() != true {
             startOpusDecoding()
-        }
+        //}
         
         // 发送命令通知设备开始解码
         let paramData = Data([BleConst.CODEC_CONTROL_DECODE_ON, BleConst.AUDIO_CHANNEL_RIGHT])
         return sendCommand(BleConst.CMD_CONTROL_CODEC, data: paramData)
     }
+     /// 获取与Flutter getExternalStorageDirectories()一致的存储目录
+    /// 在iOS上，这相当于应用的Documents目录
+    func getExternalStorageDirectory() -> URL? {
+        // 获取Documents目录，这部分与Flutter的getExternalStorageDirectories()在iOS上的行为一致
+        return FileManager.default.urls(for: .documentDirectory, in: .userDomainMask).first
+    }
     
-    /// 打开A2DP解码器
+    /// 打开A2DP解码器并开始录制
     func openA2DPDecoder() -> Bool {
-        os_log("打开A2DP解码...", log: logger, type: .info)
+         // 确保Opus解码器处于活跃状态（如果不是则启动）
+       
+        startOpusDecoding(sampleRate: 16000, channels: 2, frameSize: 1280)
+            opusProcessor?.startEncoding(sampleRate: 16000, channels: 1, frameRate: 20)
+     
+      
+        os_log("打开A2DP解码并开始录制...", log: logger, type: .info)
         let paramData = Data([BleConst.CODEC_CONTROL_A2DP_PLAY, BleConst.AUDIO_CHANNEL_RIGHT])
         return sendCommand(BleConst.CMD_CONTROL_CODEC, data: paramData)
     }
     
-    /// 关闭编解码器
+    /// 关闭编解码器并停止录制
     func closeCodec() -> Bool {
-        os_log("关闭编解码...", log: logger, type: .info)
+        os_log("关闭编解码并停止录制...", log: logger, type: .info)
+        
+        // 停止录制
+        recordingFile?.closeFile()
+        recordingFile = nil
+        
+        // 停止音频发送定时器
+        stopAudioSendTimer()
+        
+        // 清空音频缓冲区和队列
+        audioQueueLock.lock()
+        audioBuffer.removeAll()
+        audioDataQueue.removeAll()
+        audioQueueLock.unlock()
         
         // 不停止Opus解码流，只发送命令通知设备关闭编解码
         let paramData = Data([BleConst.CODEC_CONTROL_CLOSE, BleConst.AUDIO_CHANNEL_RIGHT])
@@ -1095,6 +1186,147 @@ extension BleService: SwiftOpusAudioProcessor.AudioDataCallback {
     func onAudioDataReceived(data: Data) {
         notifyAudioDataReceived(data: data)
     }
+    
+    /// 接收到解码后的PCM音频数据
+    func onAudioDataReceived1(data: Data) {
+        notifyAudioDataReceived1(data: data)
+    }
+    
+    /// 接收到编码后的音频数据
+    /// - Parameter data: 编码后的音频数据
+    func onEncodedDataReceived(data: Data) {
+        // 保存编码数据到文件
+        recordingFile?.saveAudioData(data)
+        
+        // 根据你的业务需求处理编码数据
+        // 例如：发送到蓝牙设备、保存到文件等
+        os_log("🎵 接收到编码音频数据: %d bytes", log: logger, type: .info, data.count)
+        
+        // 修复条件绑定语句
+        guard let characteristic = callAudioCharacteristic,
+              let peripheral = peripheralDevice,
+              connectionState == BleConst.STATE_CONNECTED else {
+            os_log("通话音频特征未设置或设备未连接", log: logger, type: .error)
+            return 
+        }
+      
+        // 将编码音频数据加入到队列，按每次发送120字节，每次间隔60毫秒
+        addAudioDataToSendQueue(data: data)
+        
+        // 启动音频发送定时器（如果尚未启动）
+        if !isAudioSending {
+            startAudioSendTimer()
+        }
+    }
+    
+    /**
+     * 将音频数据添加到发送队列
+     * 只有当缓冲区达到120字节时才发送数据
+     * @param data 音频数据字节数组
+     */
+    private func addAudioDataToSendQueue(data: Data) {
+        audioQueueLock.lock()
+        defer { audioQueueLock.unlock() }
+        
+        do {
+            // 将新数据添加到缓冲区
+            audioBuffer.append(contentsOf: data)
+            
+            // 当缓冲区达到120字节时，发送数据
+            while audioBuffer.count >= AUDIO_CHUNK_SIZE {
+                // 取出120字节数据
+                let chunk = Data(audioBuffer.prefix(AUDIO_CHUNK_SIZE))
+                audioBuffer.removeFirst(AUDIO_CHUNK_SIZE)
+                
+                // 将数据加入发送队列
+         
+                    audioDataQueue.append(chunk)
+               
+            }
+            
+            // os_log("音频数据已加入缓冲区，当前缓冲区大小: %d 字节", log: logger, type: .debug, audioBuffer.count)
+        } catch {
+            os_log("处理音频数据异常: %{public}@", log: logger, type: .error, error.localizedDescription)
+        }
+    }
+    
+    /**
+     * 启动音频数据发送定时器
+     */
+    private func startAudioSendTimer() {
+        guard !isAudioSending else {
+            os_log("音频发送定时器已在运行", log: logger, type: .debug)
+            return
+        }
+        
+        isAudioSending = true
+        
+        // 在主队列中创建定时器
+        DispatchQueue.main.async { [weak self] in
+            guard let self = self else { return }
+            
+            self.audioSendTimer = Timer.scheduledTimer(withTimeInterval: self.AUDIO_SEND_INTERVAL, repeats: true) { [weak self] _ in
+                self?.processAudioSendQueue()
+            }
+            
+            os_log("音频发送定时器已启动", log: self.logger, type: .info)
+        }
+    }
+    
+    /**
+     * 停止音频数据发送定时器
+     */
+    private func stopAudioSendTimer() {
+        isAudioSending = false
+        
+        DispatchQueue.main.async { [weak self] in
+            self?.audioSendTimer?.invalidate()
+            self?.audioSendTimer = nil
+            
+            guard let self = self else { return }
+            os_log("音频发送定时器已停止", log: self.logger, type: .info)
+        }
+    }
+    
+    /**
+     * 处理音频发送队列
+     */
+    private func processAudioSendQueue() {
+        audioQueueLock.lock()
+        defer { audioQueueLock.unlock() }
+        
+        guard !audioDataQueue.isEmpty else { return }
+        
+        // 从队列中取出音频数据块
+        let audioChunk = audioDataQueue.removeFirst()
+        
+        // 在后台队列中发送音频数据块
+        audioSendQueue.async { [weak self] in
+            self?.sendAudioChunk(audioChunk)
+        }
+    }
+    
+    /**
+     * 发送音频数据块到蓝牙设备
+     * @param data 要发送的音频数据
+     */
+    private func sendAudioChunk(_ data: Data) {
+        guard let characteristic = callAudioCharacteristic,
+              let peripheral = peripheralDevice,
+              connectionState == BleConst.STATE_CONNECTED else {
+            os_log("通话音频特征未设置或设备未连接，无法发送音频数据", log: logger, type: .error)
+            return
+        }
+        
+        // 发送音频数据块
+        peripheral.writeValue(data, for: characteristic, type: .withoutResponse)
+        //判断是否发送成功
+        if peripheral.state == .connected {
+            os_log("🎵 发送音频数据块: %d bytes", log: logger, type: .debug, data.count)
+        } else {
+            os_log("🎵 发送音频数据块失败: %d bytes", log: logger, type: .debug, data.count)
+        }
+    }
 }
 
 // MARK: - CBCentralManagerDelegate
@@ -1392,12 +1624,12 @@ extension BleService: CBCentralManagerDelegate {
         
         os_log("🔍 开始发现服务...", log: logger, type: .debug)
         // 查找服务
-        peripheral.discoverServices([BleConst.PRIMARY_SERVICE_UUID, BleConst.AUDIO_SERVICE_UUID])
+        peripheral.discoverServices([BleConst.PRIMARY_SERVICE_UUID, BleConst.AUDIO_SERVICE_UUID, BleConst.CALL_AUDIO_SERVICE_UUID])
         
         // 确保Opus解码器处于活跃状态
-        if opusProcessor?.isDecoding() != true {
+      //  if opusProcessor?.isDecoding() != true {
             startOpusDecoding()
-        }
+       // }
     }
     
     public func centralManager(_ central: CBCentralManager, didFailToConnect peripheral: CBPeripheral, error: Error?) {
@@ -1482,6 +1714,10 @@ extension BleService: CBPeripheralDelegate {
                 os_log("🔍 发现音频服务，开始发现音频特征...", log: logger, type: .debug)
                 // 音频服务，发现音频特征
                 peripheral.discoverCharacteristics([BleConst.RECEIVE_AUDIO_CHAR_UUID], for: service)
+            }else if service.uuid == BleConst.CALL_AUDIO_SERVICE_UUID {
+                os_log("🔍 发现通话音频服务，开始发现通话音频特征...", log: logger, type: .debug)
+                // 通话音频服务，发现通话音频特征
+                peripheral.discoverCharacteristics([BleConst.CALL_WRITE_AUDIO_CHAR_UUID], for: service)
             }
         }
     }
@@ -1513,6 +1749,10 @@ extension BleService: CBPeripheralDelegate {
                 os_log("🎵 音频特征已设置，开始发现描述符...", log: logger, type: .info)
                 // 先发现描述符，再配置通知
                 peripheral.discoverDescriptors(for: characteristic)
+            }else if characteristic.uuid == BleConst.CALL_WRITE_AUDIO_CHAR_UUID {
+                callAudioCharacteristic = characteristic
+                os_log("🎵 通话音频特征已设置，开始发现描述符...", log: logger, type: .info)
+                // 先发现描述符，再配置通知
             }
         }
         

local_plugins/ble_service/ios/ble_service/Sources/ble_service/RecordingFile.swift

@@ -0,0 +1,135 @@
+import Foundation
+import os.log
+
+/// 音频录制文件管理类
+/// 负责创建音频文件并异步写入编码后的音频数据
+@available(iOS 13.0, *)
+class RecordingFile {
+    private var currentAudioFile: URL?
+    private var fileHandle: FileHandle?
+    
+    // 用于异步写入的队列和线程
+    private let writeQueue = DispatchQueue(label: "com.yunqiinnovation.ble_service.recording_write", qos: .utility)
+    private var isWriting = false
+    private let writeLock = NSLock()
+    
+    // 用于存储音频数据的缓冲区
+    private var totalBytesWritten = 0
+    
+    var fileName = ""
+    private let logger = OSLog(subsystem: "com.yunqiinnovation.ble_service", category: "RecordingFile")
+    
+    /// 获取与Flutter getExternalStorageDirectories()一致的存储目录
+    /// 在iOS上，这相当于应用的Documents目录
+    private func getExternalStorageDirectory() -> URL? {
+        // 获取Documents目录，这部分与Flutter的getExternalStorageDirectories()在iOS上的行为一致
+        return FileManager.default.urls(for: .documentDirectory, in: .userDomainMask).first
+    }
+    
+    /// 创建新的音频文件
+    /// 如果文件已存在则直接返回
+    internal func createFile() {
+        writeLock.lock()
+        defer { writeLock.unlock() }
+        
+        if fileHandle != nil || currentAudioFile != nil {
+            return
+        }
+        
+        // 创建新的音频文件
+        let dateFormatter = DateFormatter()
+        dateFormatter.dateFormat = "yyyyMMdd_HHmmss"
+        let timestamp = dateFormatter.string(from: Date())
+        
+        // 获取与Flutter一致的外部存储目录
+        guard let externalStorageDirectory = getExternalStorageDirectory() else {
+            os_log("无法获取外部存储目录", log: logger, type: .error)
+            return
+        }
+        
+        // 使用与Android端相同的文件命名格式
+        let filePath = externalStorageDirectory.appendingPathComponent("\(fileName)_\(timestamp).opus")
+        currentAudioFile = filePath
+        
+        do {
+            // 创建文件
+            FileManager.default.createFile(atPath: filePath.path, contents: nil, attributes: nil)
+            fileHandle = try FileHandle(forWritingTo: filePath)
+            isWriting = true
+            os_log("创建文件: %@", log: logger, type: .info, filePath.path)
+        } catch {
+            os_log("创建文件失败: %@", log: logger, type: .error, error.localizedDescription)
+            currentAudioFile = nil
+            fileHandle = nil
+        }
+    }
+    
+    /// 异步保存音频数据到文件
+    /// - Parameter buffer: 编码后的音频数据
+    internal func saveAudioData(_ buffer: Data) {
+        guard isWriting, fileHandle != nil else {
+            return
+        }
+        
+        // 异步写入数据
+        writeQueue.async { [weak self] in
+            guard let self = self else { return }
+            
+            do {
+                try self.fileHandle?.write(contentsOf: buffer)
+                self.totalBytesWritten += buffer.count
+                os_log("写入音频数据: %d bytes, 总计: %d bytes", log: self.logger, type: .debug, buffer.count, self.totalBytesWritten)
+            } catch {
+                os_log("写入音频数据失败: %@", log: self.logger, type: .error, error.localizedDescription)
+            }
+        }
+    }
+    
+    /// 关闭文件并完成录制
+    internal func closeFile() {
+        writeLock.lock()
+        defer { writeLock.unlock() }
+        
+        guard let fileHandle = fileHandle, let currentFile = currentAudioFile else {
+            return
+        }
+        
+        isWriting = false
+        
+        // 等待写入队列完成
+        writeQueue.sync {
+            do {
+                try fileHandle.close()
+                
+                // 检查文件大小
+                let fileSize = try FileManager.default.attributesOfItem(atPath: currentFile.path)[.size] as? Int64 ?? 0
+                
+                if fileSize <= 0 {
+                    // 删除空文件
+                    try FileManager.default.removeItem(at: currentFile)
+                    os_log("音频文件过小已删除: %@", log: logger, type: .info, currentFile.path)
+                } else {
+                    os_log("音频文件保存完成: %@, 大小: %lld bytes", log: logger, type: .info, currentFile.path, fileSize)
+                }
+            } catch {
+                os_log("关闭文件失败: %@", log: logger, type: .error, error.localizedDescription)
+            }
+        }
+        
+        self.fileHandle = nil
+        self.currentAudioFile = nil
+        self.totalBytesWritten = 0
+    }
+    
+    /// 获取当前录制文件路径
+    /// - Returns: 当前录制文件的路径，如果没有正在录制则返回nil
+    internal func getCurrentFilePath() -> String? {
+        return currentAudioFile?.path
+    }
+    
+    /// 检查是否正在录制
+    /// - Returns: 如果正在录制返回true，否则返回false
+    internal func isRecording() -> Bool {
+        return isWriting && fileHandle != nil
+    }
+}

local_plugins/ble_service/ios/ble_service/Sources/ble_service/SwiftBleServicePlugin.swift

@@ -236,6 +236,10 @@ extension SwiftBleServicePlugin: BleService.Callback {
         }
     }
     
+    public func onAudioDataReceived1(data: Data) {
+        
+    }
+    
     public func onWakeupSignalReceived() {
         if let eventSink = statusEventSink {
             DispatchQueue.main.async {

local_plugins/ble_service/ios/ble_service/Sources/ble_service/SwiftOpusAudioProcessor.swift

@@ -1,22 +1,46 @@
 import Foundation
 import AVFoundation
-import YbridOpus
+import opus
+import JLAudioUnitKit
 
 @available(iOS 13.0, *)
 class SwiftOpusAudioProcessor: NSObject {
     
+    /// 音频数据回调协议
     protocol AudioDataCallback: AnyObject {
         func onAudioDataReceived(data: Data)
+        func onAudioDataReceived1(data: Data)
+        func onEncodedDataReceived(data: Data) // 编码数据回调
     }
     
+    // MARK: - 私有属性
+    
+    // 状态标志
     private var isDecodingActive = false
+    private var isEncodingActive = false
     private weak var callback: AudioDataCallback?
     
     // Opus 解码器相关 - 使用 YbridOpus C API
     private var decoder: OpaquePointer?
     private var sampleRate: Int32 = 16000
     private var channels: Int32 = 1
-    private let maxFrameSize: Int32 = 640  // 为16kHz优化：40ms * 16000Hz = 640 samples，与Android参数统一
+    private let maxFrameSize: Int32 = 640  // 为16kHz优化：40ms * 16000Hz = 640 samples
+    
+    // MARK: - JLOpusEncoder 编码器相关属性
+    private var jlOpusEncoder: JLOpusEncoder?
+    private var encoderFormat: JLOpusFormat?
+    
+    // MARK: - PCM数据队列相关属性
+    /// PCM数据缓存队列
+    private var pcmDataQueue = Data()
+    /// 队列访问锁，确保线程安全
+    private let queueLock = NSLock()
+    /// 每次处理的数据块大小（字节）
+    private let chunkSize = 640
+    /// 异步处理队列
+    private var processingQueue: DispatchQueue?
+    
+    // MARK: - 初始化和清理
     
     override init() {
         super.init()
@@ -26,6 +50,10 @@ class SwiftOpusAudioProcessor: NSObject {
         cleanup()
     }
     
+
+
+    // MARK: - 解码器管理
+    
     /// 线程安全的解码器初始化
     private func initializeOpusDecoder() -> Bool {
         objc_sync_enter(self)
@@ -58,12 +86,21 @@ class SwiftOpusAudioProcessor: NSObject {
         }
     }
     
+    // MARK: - 公共接口
+    
+    /// 设置音频数据回调
+    /// - Parameter callback: 回调对象
     func setCallback(_ callback: AudioDataCallback?) {
         self.callback = callback
     }
     
+    /// 启动Opus解码
+    /// - Parameters:
+    ///   - sampleRate: 采样率
+    ///   - channels: 声道数
+    ///   - frameSize: 帧大小
+    /// - Returns: 是否启动成功
     func startDecoding(sampleRate: Int32 = 16000, channels: Int32 = 1, frameSize: Int32 = 40) -> Bool {
-        // frameSize参数必须与Android版本的packetSize保持一致，确保双平台解码行为一致
         if isDecodingActive {
             return true
         }
@@ -76,10 +113,13 @@ class SwiftOpusAudioProcessor: NSObject {
         }
         
         isDecodingActive = true
+        print("Opus解码器启动成功 - 采样率: \(sampleRate)Hz, 声道: \(channels)")
         
         return true
     }
     
+    /// 停止Opus解码
+    /// - Returns: 是否停止成功
     func stopDecoding() -> Bool {
         if !isDecodingActive {
             return false
@@ -87,10 +127,13 @@ class SwiftOpusAudioProcessor: NSObject {
         
         isDecodingActive = false
         destroyOpusDecoder()
+        print("Opus解码器已停止")
         
         return true
     }
     
+    /// 处理音频数据进行解码
+    /// - Parameter data: Opus编码的音频数据
     func processAudioData(_ data: Data) {
         guard isDecodingActive, let decoder = decoder, data.count > 0 else {
             return
@@ -102,16 +145,13 @@ class SwiftOpusAudioProcessor: NSObject {
             return
         }
         
-        // 检查最小数据包大小（Opus最小包通常至少几个字节）
         guard data.count >= 3 else {
             print("音频数据包太小: \(data.count) bytes，跳过处理")
             return
         }
         
-        // 简单的 Opus 头部验证（检查是否是有效的 Opus 数据包格式）
+        // 简单的 Opus 头部验证
         let firstByte = data[0]
-        // Opus packets start with a TOC (Table of Contents) byte
-        // 检查TOC字节是否在合理范围内
         if firstByte == 0x00 || firstByte == 0xFF {
             print("可疑的Opus数据包头部: 0x\(String(format: "%02X", firstByte))，跳过处理")
             return
@@ -123,12 +163,11 @@ class SwiftOpusAudioProcessor: NSObject {
     }
     
     /// 改进的 Opus 解码方法，增强容错和线程安全
+    /// - Parameter packet: Opus数据包
     private func decodeOpusPacket(_ packet: Data) {
-        // 添加线程安全锁
         objc_sync_enter(self)
         defer { objc_sync_exit(self) }
         
-        // 验证解码器状态
         guard let decoder = decoder, isDecodingActive else {
             print("解码器未初始化或已停止")
             return
@@ -138,16 +177,13 @@ class SwiftOpusAudioProcessor: NSObject {
         var pcmBuffer = [opus_int16](repeating: 0, count: maxSamples * Int(channels))
         
         let decodedSamples = packet.withUnsafeBytes { packetPtr in
-            // 确保指针有效
             guard let uint8Ptr = packetPtr.bindMemory(to: UInt8.self).baseAddress else {
                 print("无法获取数据包指针")
                 return opus_int32(-1)
             }
             
-            // 添加异常捕获机制
             var result: opus_int32 = -1
             
-            // 使用 autoreleasepool 确保内存管理
             autoreleasepool {
                 result = opus_decode(
                     decoder,
@@ -162,12 +198,11 @@ class SwiftOpusAudioProcessor: NSObject {
             return result
         }
         
-        // 增强的错误处理：检查各种错误情况
+        // 增强的错误处理
         if decodedSamples <= 0 {
             let errorCode = decodedSamples
-            print("Opus解码失败，错误代码: \(errorCode)，重新初始化解码器")
+            print("Opus解码失败，错误代码: \(errorCode)")
             
-            // 根据错误类型进行不同的处理
             if errorCode == -1 { // OPUS_BAD_ARG
                 print("Opus解码参数错误，执行完全重置")
                 performCompleteReset()
@@ -184,18 +219,15 @@ class SwiftOpusAudioProcessor: NSObject {
             return
         }
         
-        // 验证解码结果的合理性
         guard decodedSamples <= maxFrameSize else {
             print("Opus解码样本数超出预期: \(decodedSamples) > \(maxFrameSize)")
             resetDecoder()
             return
         }
         
-        // 提取解码的音频数据
         let samplesPerChannel = Int(decodedSamples)
         let totalSamples = samplesPerChannel * Int(channels)
         
-        // 验证总样本数的合理性
         guard totalSamples <= pcmBuffer.count else {
             print("计算的总样本数超出缓冲区: \(totalSamples) > \(pcmBuffer.count)")
             resetDecoder()
@@ -209,31 +241,28 @@ class SwiftOpusAudioProcessor: NSObject {
         }
     }
     
-    // 重置解码器的简单方法
+    /// 重置解码器的简单方法
     private func resetDecoder() {
         print("执行简单解码器重置")
         destroyOpusDecoder()
         _ = initializeOpusDecoder()
     }
     
-    // 执行完全重置（包括停止和重启解码）
+    /// 执行完全重置（包括停止和重启解码）
     private func performCompleteReset() {
         print("执行完全解码器重置")
         let wasActive = isDecodingActive
         let currentSampleRate = self.sampleRate
         let currentChannels = self.channels
         
-        // 完全停止解码
         if wasActive {
             isDecodingActive = false
             destroyOpusDecoder()
         }
         
-        // 重新初始化所有状态
         self.sampleRate = currentSampleRate
         self.channels = currentChannels
         
-        // 如果之前是活跃状态，重新启动
         if wasActive {
             guard initializeOpusDecoder() else {
                 print("完全重置后重新初始化失败")
@@ -243,29 +272,312 @@ class SwiftOpusAudioProcessor: NSObject {
             print("完全重置成功，解码器已重新启动")
         }
     }
-    
+  
+    /// 处理解码后的PCM音频数据
+    /// - Parameter pcmData: 解码后的PCM数据数组
     private func processPCMData(_ pcmData: [opus_int16]) {
-        // 转换为 Data
         let data = Data(bytes: pcmData, count: pcmData.count * MemoryLayout<opus_int16>.size)
         
-        // 回调通知
+        guard !data.isEmpty else {
+            print("PCM数据为空")
+            return
+        }
+        
+        print("Opus解码数据: \(data.count) bytes, 声道数: \(channels)")
+        
+        if channels == 2 {
+            processStereoAudioData(data)
+        } else if channels == 1 {
+            DispatchQueue.main.async { [weak self] in
+                self?.callback?.onAudioDataReceived(data: data)
+            }
+        } else {
+            print("不支持的声道数: \(channels)")
+        }
+    }
+    
+    /// 处理双声道音频数据，分离左右声道
+    /// - Parameter stereoData: 交错的双声道PCM数据
+    private func processStereoAudioData(_ stereoData: Data) {
+        let sampleCount = stereoData.count / 4 // 每个样本4字节（左右声道各2字节）
+        
+        guard sampleCount > 0 else {
+            print("双声道数据样本数为0")
+            return
+        }
+        
+        var leftBuffer = Data()
+        var rightBuffer = Data()
+        
+        leftBuffer.reserveCapacity(sampleCount * 2)
+        rightBuffer.reserveCapacity(sampleCount * 2)
+        
+        stereoData.withUnsafeBytes { bytes in
+            let uint8Ptr = bytes.bindMemory(to: UInt8.self)
+            
+            for i in 0..<sampleCount {
+                let stereoIndex = i * 4
+                
+                // 左声道（低位字节在前，高位字节在后）
+                leftBuffer.append(uint8Ptr[stereoIndex])
+                leftBuffer.append(uint8Ptr[stereoIndex + 1])
+                
+                // 右声道
+                rightBuffer.append(uint8Ptr[stereoIndex + 2])
+                rightBuffer.append(uint8Ptr[stereoIndex + 3])
+            }
+        }
+        
+        print("分离音频数据 - 左声道: \(leftBuffer.count) bytes, 右声道: \(rightBuffer.count) bytes")
+        
+        // 使用SpeexKit编码右声道数据
+        //encodePCMData(rightBuffer)
+        
+        // 回调分离后的音频数据
         DispatchQueue.main.async { [weak self] in
-            self?.callback?.onAudioDataReceived(data: data)
+            self?.callback?.onAudioDataReceived(data: rightBuffer)
+            self?.callback?.onAudioDataReceived1(data: leftBuffer)
         }
     }
     
+  
+    // MARK: - 状态查询方法
+    
+    /// 获取解码器状态
+    /// - Returns: 是否正在解码
     func isDecoding() -> Bool {
         return isDecodingActive
     }
     
-    func cleanup() {
+    /// 获取编码器状态
-        stopDecoding()
+    /// - Returns: 是否正在编码
+    func isEncoding() -> Bool {
+        return isEncodingActive
     }
     
-    // MARK: - 公共接口
-    
     /// 获取当前配置信息
+    /// - Returns: 采样率和声道数
     func getCurrentConfig() -> (sampleRate: Int32, channels: Int32) {
         return (sampleRate, channels)
     }
+    
+    /// 获取编码器配置信息
+    /// - Returns: 编码器配置元组
+    func getEncoderConfig() -> (sampleRate: Int32, channels: Int32) {
+        return (sampleRate, channels)
+    }
+    
+    
+    /// 启动JLOpusEncoder编码（完善版本）
+    /// - Parameters:
+    ///   - sampleRate: 采样率（默认16000Hz）
+    ///   - channels: 声道数（默认1）
+    ///   - frameRate: 帧长度（默认20ms）
+    ///   - isHeader: 是否包含头部信息
+    /// - Returns: 是否启动成功
+    func startEncoding(sampleRate: Int32 = 16000, 
+                      channels: Int32 = 1, 
+                      frameRate: Int32 = 20,
+                      isHeader: Bool = false) -> Bool {
+        if isEncodingActive {
+            return true
+        }
+        
+        print("启动JLOpusEncoder编码 - 采样率: \(sampleRate)Hz, 声道: \(channels), 帧长度: \(frameRate)ms, 头部: \(isHeader)")
+        
+        // 创建JLOpusFormat配置
+        encoderFormat = JLOpusFormat.defaultFormats()
+        encoderFormat?.sampleRate = Int32(sampleRate)
+        encoderFormat?.channels = Int32(channels)
+        encoderFormat?.frameDuration = Int32(frameRate)
+        encoderFormat?.hasDataHeader = isHeader
+        
+        // 根据配置计算数据大小
+        if channels == 1 {
+            encoderFormat?.dataSize = frameRate == 20 ? 40 : 80  // 20ms=40bytes, 40ms=80bytes
+        } else {
+            encoderFormat?.dataSize = frameRate == 20 ? 80 : 160  // 双声道翻倍
+        }
+        
+        guard let format = encoderFormat else {
+            print("编码器格式配置失败")
+            return false
+        }
+        
+        // 初始化JLOpusEncoder
+        jlOpusEncoder = JLOpusEncoder(format: format, delegate: self)
+        
+        guard jlOpusEncoder != nil else {
+            print("JLOpusEncoder初始化失败")
+            return false
+        }
+        
+        isEncodingActive = true
+        print("JLOpusEncoder启动成功")
+        
+        return true
+    }
+    
+    /// 停止JLOpusEncoder编码（完善版本）
+    /// - Returns: 是否停止成功
+    func stopEncoding() -> Bool {
+        if !isEncodingActive {
+            return false
+        }
+        
+        isEncodingActive = false
+        
+        // 清空PCM数据队列
+        clearPCMQueue()
+        
+        // 释放JLOpusEncoder资源
+        jlOpusEncoder?.opusOnRelease()
+        jlOpusEncoder = nil
+        encoderFormat = nil
+        processingQueue = nil
+        
+        print("JLOpusEncoder已停止")
+        
+        return true
+    }
+    
+    /// 编码PCM音频数据
+    /// - Parameter pcmData: PCM音频数据
+    func encodePCMData(_ pcmData: Data) {
+        guard isEncodingActive, let encoder = jlOpusEncoder else {
+            print("编码器未启动或不可用")
+            return
+        }
+        
+        guard !pcmData.isEmpty else {
+            print("PCM数据为空")
+            return
+        }
+        
+        print("接收PCM数据 - 数据大小: \(pcmData.count) bytes")
+        
+        // 将新数据加入队列
+        queueLock.lock()
+        pcmDataQueue.append(pcmData)
+        let currentQueueSize = pcmDataQueue.count
+        queueLock.unlock()
+        
+        print("PCM数据已加入队列 - 当前队列大小: \(currentQueueSize) bytes")
+        
+        // 异步处理队列中的数据
+        processQueuedData()
+    }
+    
+    /// 处理队列中的PCM数据
+    private func processQueuedData() {
+        guard isEncodingActive, let encoder = jlOpusEncoder else {
+            return
+        }
+        
+        // 使用专用队列进行异步处理
+        if processingQueue == nil {
+            processingQueue = DispatchQueue(label: "com.opus.processing", qos: .userInitiated)
+        }
+        
+        processingQueue?.async { [weak self, weak encoder] in
+            guard let self = self, let encoder = encoder else {
+                print("编码器或处理器已释放")
+                return
+            }
+            
+            while self.isEncodingActive {
+                var chunkData: Data?
+                
+                // 从队列中取出数据
+                self.queueLock.lock()
+                if self.pcmDataQueue.count >= self.chunkSize {
+                    // 取出640字节数据
+                    chunkData = self.pcmDataQueue.subdata(in: 0..<self.chunkSize)
+                    self.pcmDataQueue.removeSubrange(0..<self.chunkSize)
+                } else if self.pcmDataQueue.count > 0 {
+                    // 不足640字节，补齐静音数据
+                    let availableBytes = self.pcmDataQueue.count
+                    let silentBytesNeeded = self.chunkSize - availableBytes
+                    
+                    // 创建静音数据（16位PCM静音为0）
+                    let silentData = Data(repeating: 0, count: silentBytesNeeded)
+                    
+                    // 组合实际数据和静音数据
+                    chunkData = Data()
+                    chunkData?.append(self.pcmDataQueue)
+                    chunkData?.append(silentData)
+                    
+                    // 清空队列
+                    self.pcmDataQueue.removeAll()
+                    
+                    print("数据不足640字节，已补齐静音数据 - 实际数据: \(availableBytes) bytes, 静音数据: \(silentBytesNeeded) bytes")
+                }
+                self.queueLock.unlock()
+                
+                // 如果有数据需要编码
+                if let dataToEncode = chunkData {
+                    guard self.isEncodingActive else {
+                        print("编码已停止，退出处理循环")
+                        break
+                    }
+                    
+                    // 编码数据块
+                    encoder.opusEncode(dataToEncode)
+                    print("已编码音频块 - 数据大小: \(dataToEncode.count) bytes")
+                } else {
+                    // 队列为空，退出循环
+                    break
+                }
+            }
+            
+            print("队列处理完成")
+        }
+    }
+    
+    /// 清空PCM数据队列
+    private func clearPCMQueue() {
+        queueLock.lock()
+        pcmDataQueue.removeAll()
+        queueLock.unlock()
+        print("PCM数据队列已清空")
+    }
+    
+    
+    /// 清理所有资源
+    func cleanup() {
+        stopDecoding()
+        stopEncoding()
+    }
+}
+
+// MARK: - JLOpusEncoderDelegate
+@available(iOS 13.0, *)
+extension SwiftOpusAudioProcessor: JLOpusEncoderDelegate {
+    
+    /// JLOpusEncoder编码回调
+    /// - Parameters:
+    ///   - encoder: 编码器实例
+    ///   - data: 编码后的Opus数据
+    ///   - error: 错误信息
+    func opusEncoder(_ encoder: JLOpusEncoder, data: Data?, error: (any Error)?) {
+        if let error = error {
+            print("Opus编码错误: \(error.localizedDescription)")
+            return
+        }
+        
+        guard let opusData = data, !opusData.isEmpty else {
+            print("编码后的Opus数据为空")
+            return
+        }
+        
+        print("Opus编码成功 - 数据大小: \(opusData.count) bytes")
+        
+        // 回调编码后的数据到主线程
+        DispatchQueue.main.async { [weak self] in
+            self?.callback?.onEncodedDataReceived(data: opusData)
+        }
+    }
 }
+    
+   
+

local_plugins/ble_service/ios/ble_service/Sources/ble_service/opuse.swift

@@ -0,0 +1,85 @@
+import Foundation
+import opus
+
+class OpusAudioProcessor {
+    private var encoder: OpaquePointer?
+    private var decoder: OpaquePointer?
+    
+    /**
+     * 初始化Opus编码器
+     * @param sampleRate 采样率 (8000, 12000, 16000, 24000, 48000)
+     * @param channels 声道数 (1 = 单声道, 2 = 立体声)
+     * @param application 应用类型 (OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+     */
+    func initializeEncoder(sampleRate: Int32, channels: Int32, application: Int32) -> Bool {
+        var error: Int32 = 0
+        encoder = opus_encoder_create(sampleRate, channels, application, &error)
+        return error == OPUS_OK
+    }
+    
+    /**
+     * 初始化Opus解码器
+     * @param sampleRate 采样率
+     * @param channels 声道数
+     */
+    func initializeDecoder(sampleRate: Int32, channels: Int32) -> Bool {
+        var error: Int32 = 0
+        decoder = opus_decoder_create(sampleRate, channels, &error)
+        return error == OPUS_OK
+    }
+    
+    /**
+     * 编码音频数据
+     * @param pcmData PCM音频数据
+     * @param frameSize 帧大小
+     * @param maxDataBytes 最大输出字节数
+     * @return 编码后的数据
+     */
+    func encode(pcmData: [Int16], frameSize: Int32, maxDataBytes: Int32) -> Data? {
+        guard let encoder = encoder else { return nil }
+        
+        let outputBuffer = UnsafeMutablePointer<UInt8>.allocate(capacity: Int(maxDataBytes))
+        defer { outputBuffer.deallocate() }
+        
+        let encodedBytes = opus_encode(encoder, pcmData, frameSize, outputBuffer, maxDataBytes)
+        
+        if encodedBytes > 0 {
+            return Data(bytes: outputBuffer, count: Int(encodedBytes))
+        }
+        return nil
+    }
+    
+    /**
+     * 解码音频数据
+     * @param encodedData 编码后的数据
+     * @param frameSize 帧大小
+     * @return 解码后的PCM数据
+     */
+    func decode(encodedData: Data, frameSize: Int32) -> [Int16]? {
+        guard let decoder = decoder else { return nil }
+        
+        let outputBuffer = UnsafeMutablePointer<Int16>.allocate(capacity: Int(frameSize * 2)) // 假设最多2声道
+        defer { outputBuffer.deallocate() }
+        
+        let decodedSamples = encodedData.withUnsafeBytes { bytes in
+            opus_decode(decoder, bytes.bindMemory(to: UInt8.self).baseAddress, Int32(encodedData.count), outputBuffer, frameSize, 0)
+        }
+        
+        if decodedSamples > 0 {
+            return Array(UnsafeBufferPointer(start: outputBuffer, count: Int(decodedSamples * 2)))
+        }
+        return nil
+    }
+ 
+    /**
+     * 清理资源
+     */
+    deinit {
+        if let encoder = encoder {
+            opus_encoder_destroy(encoder)
+        }
+        if let decoder = decoder {
+            opus_decoder_destroy(decoder)
+        }
+    }
+}

local_plugins/ble_service/ios/ble_service/opus.xcframework/Info.plist

@@ -0,0 +1,43 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>AvailableLibraries</key>
+	<array>
+		<dict>
+			<key>BinaryPath</key>
+			<string>opus.framework/opus</string>
+			<key>LibraryIdentifier</key>
+			<string>ios-x86_64-simulator</string>
+			<key>LibraryPath</key>
+			<string>opus.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>x86_64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+			<key>SupportedPlatformVariant</key>
+			<string>simulator</string>
+		</dict>
+		<dict>
+			<key>BinaryPath</key>
+			<string>opus.framework/opus</string>
+			<key>LibraryIdentifier</key>
+			<string>ios-arm64</string>
+			<key>LibraryPath</key>
+			<string>opus.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>arm64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+		</dict>
+	</array>
+	<key>CFBundlePackageType</key>
+	<string>XFWK</string>
+	<key>XCFrameworkFormatVersion</key>
+	<string>1.0</string>
+</dict>
+</plist>

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-arm64/opus.framework/Headers/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-arm64/opus.framework/Headers/opus_defines.h

@@ -0,0 +1,753 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_VARIABLE</dt><dd>Optimize the frame size dynamically.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_VARIABLE</dt><dd>Optimize the frame size dynamically.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-arm64/opus.framework/Headers/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-arm64/opus.framework/Headers/opus_types.h

@@ -0,0 +1,159 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+#endif  /* OPUS_TYPES_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-arm64/opus.framework/Modules/module.modulemap

@@ -0,0 +1,6 @@
+framework module opus {
+  umbrella header "opus.h"
+
+  export *
+  module * { export * }
+}

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-x86_64-simulator/opus.framework/Headers/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-x86_64-simulator/opus.framework/Headers/opus_defines.h

@@ -0,0 +1,753 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_VARIABLE</dt><dd>Optimize the frame size dynamically.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_VARIABLE</dt><dd>Optimize the frame size dynamically.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-x86_64-simulator/opus.framework/Headers/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-x86_64-simulator/opus.framework/Headers/opus_types.h

@@ -0,0 +1,159 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+#endif  /* OPUS_TYPES_H */

local_plugins/ble_service/ios/ble_service/opus.xcframework/ios-x86_64-simulator/opus.framework/Modules/module.modulemap

@@ -0,0 +1,6 @@
+framework module opus {
+  umbrella header "opus.h"
+
+  export *
+  module * { export * }
+}

local_plugins/chat_api/ios/chat_api/Package.swift

@@ -13,7 +13,7 @@ let package = Package(
         .library(name: "chat-api", targets: ["chat_api"])
     ],
     dependencies: [
-        .package(url: "https://github.com/MacPaw/OpenAI.git", from: "0.4.3"),
+        .package(url: "https://github.com/MacPaw/OpenAI.git", exact: "0.4.3"),
         .package(url: "https://github.com/modelcontextprotocol/swift-sdk.git", from: "0.9.0"),
         .package(path: "../../../location_service/ios/location_service"),
         .package(path: "../../../qq_music/ios/qq_music"),