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快速入门：创建实时语音代理

按照以下快速入门，或者使用基于浏览器的语音用户界面获取一个完全功能正常的Web应用程序：

语音直播通用助理示例

本文介绍如何在 Microsoft Foundry 门户中的 Foundry 工具中结合生成式AI，使用“Voice Live”与 Azure 语音技术。

创建并运行一个应用程序，将 Voice Live 直接与生成型 AI 模型结合使用，以创建实时语音代理。

  • 使用模型直接允许为每个会话指定自定义指令（提示），为动态或实验用例提供更大的灵活性。

  • 如果希望对会话参数进行精细控制，或者需要频繁调整提示或配置，而无需在门户中更新代理，则模型可能更可取。

  • 基于模型的会话的代码在某些方面更简单，因为它不需要管理代理 ID 或特定于代理的设置。

  • 直接模型使用适用于不需要代理级抽象或内置逻辑的方案。

若要改用语音实时 API 和代理，请参阅 语音实时 API 代理快速入门。

先决条件

小窍门

若要使用 Voice Live，无需使用 Microsoft Foundry 资源部署音频模型。 Voice Live 是全托管的，模型会自动为你部署。 有关模型可用性的详细信息，请参阅 Voice Live 概述文档。

在语音实验室中试用实时语音

若要试用 Voice Live 演示，请执行以下步骤：

  1. 登录到 Microsoft Foundry。 确保 New Foundry 开关处于打开状态。 这些步骤适用于 Foundry（新）。
  2. 从右上角菜单中选择“ 生成 ”。
  3. 在左窗格中选择 “模型 ”。
  4. AI Services 选项卡显示可在 Foundry 门户中现用的 Azure AI 模型。 选择Azure语音 - 语音直播打开语音直播场。
  5. 使用下拉菜单选择方案和语音。 （可选）配置语音代理行为的其他参数。 例如， 主动参与 切换允许代理首先在对话中说话。
  6. 准备就绪后，选择 “开始” 以使用设备的麦克风和扬声器与语音代理聊天。
  7. 选择 “结束 ”以结束聊天会话。

其他 Foundry（新版）功能

Foundry（新增）门户中提供了以下语音功能：

本文介绍如何使用 voiceLive SDK for Python 将 Voice Live 与 Microsoft Foundry 模型配合使用。

参考文档 | 软件包（PyPi） | GitHub 上的其他示例

创建并运行一个应用程序，将 Voice Live 直接与生成型 AI 模型结合使用，以创建实时语音代理。

  • 使用模型直接允许为每个会话指定自定义指令（提示），为动态或实验用例提供更大的灵活性。

  • 如果希望对会话参数进行精细控制，或者需要频繁调整提示或配置，而无需在门户中更新代理，则模型可能更可取。

  • 基于模型的会话的代码在某些方面更简单，因为它不需要管理代理 ID 或特定于代理的设置。

  • 直接模型使用适用于不需要代理级抽象或内置逻辑的方案。

若要改用语音实时 API 和代理，请参阅 语音实时 API 代理快速入门。

先决条件

小窍门

若要使用 Voice Live，无需使用 Microsoft Foundry 资源部署音频模型。 Voice Live 是全托管的，模型会自动为你部署。 有关模型可用性的详细信息，请参阅 Voice Live 概述文档。

Microsoft Entra ID先决条件

若要使用 Microsoft Entra ID 进行推荐的无密钥身份验证，需要：

  • 安装用于无密钥身份验证的 Azure CLI 和 Microsoft Entra ID。
  • 将Cognitive Services User角色分配给用户帐户。 可以在 Azure 门户中的 访问控制（IAM）>添加角色分配下分配角色。

设置

  1. 创建新文件夹 voice-live-quickstart，并使用以下命令转到快速入门文件夹：

    mkdir voice-live-quickstart && cd voice-live-quickstart

  2. 创建虚拟环境。 如果已安装 Python 3.10 或更高版本，可以使用以下命令创建虚拟环境：

    py -3 -m venv .venv
.venv\scripts\activate

    激活Python环境意味着从命令行运行 python 或 pip 时，请使用应用程序 .venv 文件夹中包含的Python解释器。 可以使用 deactivate 命令退出 python 虚拟环境，并在需要时重新激活它。

    小窍门

    建议创建并激活新的Python环境，以用于安装本教程所需的包。 请勿将包安装到你的全局 Python 安装中。 安装python包时，应始终使用虚拟或 conda 环境，否则可能会中断Python的全局安装。

  3. 创建名为 requirements.txt的文件。 将以下包添加到文件：

    azure-ai-voicelive[aiohttp]
pyaudio
python-dotenv
azure-identity

  4. 安装这些软件包：

    pip install -r requirements.txt

检索资源信息

在要在其中运行代码的文件夹中创建一 .env 个名为的新文件。

在 .env 文件中，添加以下用于身份验证的环境变量：

AZURE_VOICELIVE_ENDPOINT=<your_endpoint>
AZURE_VOICELIVE_MODEL=<your_model>
AZURE_VOICELIVE_API_VERSION=2025-10-01
AZURE_VOICELIVE_API_KEY=<your_api_key> # Only required if using API key authentication

将默认值替换为实际终结点、模型、API 版本和 API 密钥。

变量名称 价值
AZURE_VOICELIVE_ENDPOINT 从 Azure 门户检查资源时，可以在 Keys 和 Endpoint 节中找到此值。
AZURE_VOICELIVE_MODEL 要使用的模型。 例如，gpt-4o 或 gpt-realtime-mini。 有关模型可用性的详细信息，请参阅 Voice Live API 概述文档。
AZURE_VOICELIVE_API_VERSION 要使用的 API 版本。 例如，2025-10-01。

详细了解无密钥身份验证，以及如何设置环境变量。

启动会话

本快速入门中的示例代码使用Microsoft Entra ID或 API 密钥进行身份验证。 可以将脚本参数设置为 API 密钥或访问令牌。

  1. 使用以下代码创建 voice-live-quickstart.py 文件：

    # -------------------------------------------------------------------------
# Copyright (c) Microsoft Corporation. All rights reserved.
# Licensed under the MIT License.
# -------------------------------------------------------------------------
from __future__ import annotations
import os
import sys
import argparse
import asyncio
import base64
from datetime import datetime
import logging
import queue
import signal
from typing import Union, Optional, TYPE_CHECKING, cast

from azure.core.credentials import AzureKeyCredential
from azure.core.credentials_async import AsyncTokenCredential
from azure.identity.aio import AzureCliCredential, DefaultAzureCredential

from azure.ai.voicelive.aio import connect
from azure.ai.voicelive.models import (
    AudioEchoCancellation,
    AudioNoiseReduction,
    AzureStandardVoice,
    InputAudioFormat,
    Modality,
    OutputAudioFormat,
    RequestSession,
    ServerEventType,
    ServerVad
)
from dotenv import load_dotenv
import pyaudio

if TYPE_CHECKING:
    # Only needed for type checking; avoids runtime import issues
    from azure.ai.voicelive.aio import VoiceLiveConnection

## Change to the directory where this script is located
os.chdir(os.path.dirname(os.path.abspath(__file__)))

# Environment variable loading
load_dotenv('./.env', override=True)

# Set up logging
## Add folder for logging
if not os.path.exists('logs'):
    os.makedirs('logs')

## Add timestamp for logfiles
timestamp = datetime.now().strftime("%Y-%m-%d_%H-%M-%S")

## Set up logging
logging.basicConfig(
    filename=f'logs/{timestamp}_voicelive.log',
    filemode="w",
    format='%(asctime)s:%(name)s:%(levelname)s:%(message)s',
    level=logging.INFO
)
logger = logging.getLogger(__name__)

class AudioProcessor:
    """
    Handles real-time audio capture and playback for the voice assistant.

    Threading Architecture:
    - Main thread: Event loop and UI
    - Capture thread: PyAudio input stream reading
    - Send thread: Async audio data transmission to VoiceLive
    - Playback thread: PyAudio output stream writing
    """

    loop: asyncio.AbstractEventLoop

    class AudioPlaybackPacket:
        """Represents a packet that can be sent to the audio playback queue."""
        def __init__(self, seq_num: int, data: Optional[bytes]):
            self.seq_num = seq_num
            self.data = data

    def __init__(self, connection):
        self.connection = connection
        self.audio = pyaudio.PyAudio()

        # Audio configuration - PCM16, 24kHz, mono as specified
        self.format = pyaudio.paInt16
        self.channels = 1
        self.rate = 24000
        self.chunk_size = 1200 # 50ms

        # Capture and playback state
        self.input_stream = None

        self.playback_queue: queue.Queue[AudioProcessor.AudioPlaybackPacket] = queue.Queue()
        self.playback_base = 0
        self.next_seq_num = 0
        self.output_stream: Optional[pyaudio.Stream] = None

        logger.info("AudioProcessor initialized with 24kHz PCM16 mono audio")

    def start_capture(self):
        """Start capturing audio from microphone."""
        def _capture_callback(
            in_data,      # data
            _frame_count,  # number of frames
            _time_info,    # dictionary
            _status_flags):
            """Audio capture thread - runs in background."""
            audio_base64 = base64.b64encode(in_data).decode("utf-8")
            asyncio.run_coroutine_threadsafe(
                self.connection.input_audio_buffer.append(audio=audio_base64), self.loop
            )
            return (None, pyaudio.paContinue)

        if self.input_stream:
            return

        # Store the current event loop for use in threads
        self.loop = asyncio.get_event_loop()

        try:
            self.input_stream = self.audio.open(
                format=self.format,
                channels=self.channels,
                rate=self.rate,
                input=True,
                frames_per_buffer=self.chunk_size,
                stream_callback=_capture_callback,
            )
            logger.info("Started audio capture")

        except Exception:
            logger.exception("Failed to start audio capture")
            raise

    def start_playback(self):
        """Initialize audio playback system."""
        if self.output_stream:
            return

        remaining = bytes()
        def _playback_callback(
            _in_data,
            frame_count,  # number of frames
            _time_info,
            _status_flags):

            nonlocal remaining
            frame_count *= pyaudio.get_sample_size(pyaudio.paInt16)

            out = remaining[:frame_count]
            remaining = remaining[frame_count:]

            while len(out) < frame_count:
                try:
                    packet = self.playback_queue.get_nowait()
                except queue.Empty:
                    out = out + bytes(frame_count - len(out))
                    continue
                except Exception:
                    logger.exception("Error in audio playback")
                    raise

                if not packet or not packet.data:
                    # None packet indicates end of stream
                    logger.info("End of playback queue.")
                    break

                if packet.seq_num < self.playback_base:
                    # skip requested
                    # ignore skipped packet and clear remaining
                    if len(remaining) > 0:
                        remaining = bytes()
                    continue

                num_to_take = frame_count - len(out)
                out = out + packet.data[:num_to_take]
                remaining = packet.data[num_to_take:]

            if len(out) >= frame_count:
                return (out, pyaudio.paContinue)
            else:
                return (out, pyaudio.paComplete)

        try:
            self.output_stream = self.audio.open(
                format=self.format,
                channels=self.channels,
                rate=self.rate,
                output=True,
                frames_per_buffer=self.chunk_size,
                stream_callback=_playback_callback
            )
            logger.info("Audio playback system ready")
        except Exception:
            logger.exception("Failed to initialize audio playback")
            raise

    def _get_and_increase_seq_num(self):
        seq = self.next_seq_num
        self.next_seq_num += 1
        return seq

    def queue_audio(self, audio_data: Optional[bytes]) -> None:
        """Queue audio data for playback."""
        self.playback_queue.put(
            AudioProcessor.AudioPlaybackPacket(
                seq_num=self._get_and_increase_seq_num(),
                data=audio_data))

    def skip_pending_audio(self):
        """Skip current audio in playback queue."""
        self.playback_base = self._get_and_increase_seq_num()

    def shutdown(self):
        """Clean up audio resources."""
        if self.input_stream:
            self.input_stream.stop_stream()
            self.input_stream.close()
            self.input_stream = None

        logger.info("Stopped audio capture")

        # Inform thread to complete
        if self.output_stream:
            self.skip_pending_audio()
            self.queue_audio(None)
            self.output_stream.stop_stream()
            self.output_stream.close()
            self.output_stream = None

        logger.info("Stopped audio playback")

        if self.audio:
            self.audio.terminate()

        logger.info("Audio processor cleaned up")

class BasicVoiceAssistant:
    """Basic voice assistant implementing the VoiceLive SDK patterns."""

    def __init__(
        self,
        endpoint: str,
        credential: Union[AzureKeyCredential, AsyncTokenCredential],
        model: str,
        voice: str,
        instructions: str,
    ):

        self.endpoint = endpoint
        self.credential = credential
        self.model = model
        self.voice = voice
        self.instructions = instructions
        self.connection: Optional["VoiceLiveConnection"] = None
        self.audio_processor: Optional[AudioProcessor] = None
        self.session_ready = False
        self._active_response = False
        self._response_api_done = False

    async def start(self):
        """Start the voice assistant session."""
        try:
            logger.info("Connecting to VoiceLive API with model %s", self.model)

            # Connect to VoiceLive WebSocket API
            async with connect(
                endpoint=self.endpoint,
                credential=self.credential,
                model=self.model,
            ) as connection:
                conn = connection
                self.connection = conn

                # Initialize audio processor
                ap = AudioProcessor(conn)
                self.audio_processor = ap

                # Configure session for voice conversation
                await self._setup_session()

                # Start audio systems
                ap.start_playback()

                logger.info("Voice assistant ready! Start speaking...")
                print("\n" + "=" * 60)
                print("🎤 VOICE ASSISTANT READY")
                print("Start speaking to begin conversation")
                print("Press Ctrl+C to exit")
                print("=" * 60 + "\n")

                # Process events
                await self._process_events()
        finally:
            if self.audio_processor:
                self.audio_processor.shutdown()

    async def _setup_session(self):
        """Configure the VoiceLive session for audio conversation."""
        logger.info("Setting up voice conversation session...")

        # Create voice configuration
        voice_config: Union[AzureStandardVoice, str]
        if self.voice.startswith("en-US-") or self.voice.startswith("en-CA-") or "-" in self.voice:
            # Azure voice
            voice_config = AzureStandardVoice(name=self.voice)
        else:
            # OpenAI voice (alloy, echo, fable, onyx, nova, shimmer)
            voice_config = self.voice

        # Create turn detection configuration
        turn_detection_config = ServerVad(
            threshold=0.5,
            prefix_padding_ms=300,
            silence_duration_ms=500)

        # Create session configuration
        session_config = RequestSession(
            modalities=[Modality.TEXT, Modality.AUDIO],
            instructions=self.instructions,
            voice=voice_config,
            input_audio_format=InputAudioFormat.PCM16,
            output_audio_format=OutputAudioFormat.PCM16,
            turn_detection=turn_detection_config,
            input_audio_echo_cancellation=AudioEchoCancellation(),
            input_audio_noise_reduction=AudioNoiseReduction(type="azure_deep_noise_suppression"),
        )

        conn = self.connection
        assert conn is not None, "Connection must be established before setting up session"
        await conn.session.update(session=session_config)

        logger.info("Session configuration sent")

    async def _process_events(self):
        """Process events from the VoiceLive connection."""
        try:
            conn = self.connection
            assert conn is not None, "Connection must be established before processing events"
            async for event in conn:
                await self._handle_event(event)
        except Exception:
            logger.exception("Error processing events")
            raise

    async def _handle_event(self, event):
        """Handle different types of events from VoiceLive."""
        logger.debug("Received event: %s", event.type)
        ap = self.audio_processor
        conn = self.connection
        assert ap is not None, "AudioProcessor must be initialized"
        assert conn is not None, "Connection must be established"

        if event.type == ServerEventType.SESSION_UPDATED:
            logger.info("Session ready: %s", event.session.id)
            self.session_ready = True

            # Start audio capture once session is ready
            ap.start_capture()

        elif event.type == ServerEventType.INPUT_AUDIO_BUFFER_SPEECH_STARTED:
            logger.info("User started speaking - stopping playback")
            print("🎤 Listening...")

            ap.skip_pending_audio()

        elif event.type == ServerEventType.INPUT_AUDIO_BUFFER_SPEECH_STOPPED:
            logger.info("🎤 User stopped speaking")
            print("🤔 Processing...")

        elif event.type == ServerEventType.RESPONSE_CREATED:
            logger.info("🤖 Assistant response created")
            self._active_response = True
            self._response_api_done = False

        elif event.type == ServerEventType.RESPONSE_AUDIO_DELTA:
            logger.debug("Received audio delta")
            ap.queue_audio(event.delta)

        elif event.type == ServerEventType.RESPONSE_AUDIO_DONE:
            logger.info("🤖 Assistant finished speaking")
            print("🎤 Ready for next input...")

        elif event.type == ServerEventType.RESPONSE_DONE:
            logger.info("✅ Response complete")
            self._active_response = False
            self._response_api_done = True

        elif event.type == ServerEventType.ERROR:
            msg = event.error.message
            if "Cancellation failed: no active response" in msg:
                logger.debug("Benign cancellation error: %s", msg)
            else:
                logger.error("❌ VoiceLive error: %s", msg)
                print(f"Error: {msg}")

        elif event.type == ServerEventType.CONVERSATION_ITEM_CREATED:
            logger.debug("Conversation item created: %s", event.item.id)

        else:
            logger.debug("Unhandled event type: %s", event.type)


def parse_arguments():
    """Parse command line arguments."""
    parser = argparse.ArgumentParser(
        description="Basic Voice Assistant using Azure VoiceLive SDK",
        formatter_class=argparse.ArgumentDefaultsHelpFormatter,
    )

    parser.add_argument(
        "--api-key",
        help="Azure VoiceLive API key. If not provided, will use AZURE_VOICELIVE_API_KEY environment variable.",
        type=str,
        default=os.environ.get("AZURE_VOICELIVE_API_KEY"),
    )

    parser.add_argument(
        "--endpoint",
        help="Azure VoiceLive endpoint",
        type=str,
        default=os.environ.get("AZURE_VOICELIVE_ENDPOINT", "https://your-resource-name.services.ai.azure.com/"),
    )

    parser.add_argument(
        "--model",
        help="VoiceLive model to use",
        type=str,
        default=os.environ.get("AZURE_VOICELIVE_MODEL", "gpt-realtime"),
    )

    parser.add_argument(
        "--voice",
        help="Voice to use for the assistant. E.g. alloy, echo, fable, en-US-AvaNeural, en-US-GuyNeural",
        type=str,
        default=os.environ.get("AZURE_VOICELIVE_VOICE", "en-US-Ava:DragonHDLatestNeural"),
    )

    parser.add_argument(
        "--instructions",
        help="System instructions for the AI assistant",
        type=str,
        default=os.environ.get(
            "AZURE_VOICELIVE_INSTRUCTIONS",
            "You are a helpful AI assistant. Respond naturally and conversationally. "
            "Keep your responses concise but engaging.",
        ),
    )

    parser.add_argument(
        "--use-token-credential", help="Use Azure token credential instead of API key", action="store_true", default=False
    )

    parser.add_argument("--verbose", help="Enable verbose logging", action="store_true")

    return parser.parse_args()


def main():
    """Main function."""
    args = parse_arguments()

    # Set logging level
    if args.verbose:
        logging.getLogger().setLevel(logging.DEBUG)

    # Validate credentials
    if not args.api_key and not args.use_token_credential:
        print("❌ Error: No authentication provided")
        print("Please provide an API key using --api-key or set AZURE_VOICELIVE_API_KEY environment variable,")
        print("or use --use-token-credential for Azure authentication.")
        sys.exit(1)

    # Create client with appropriate credential
    credential: Union[AzureKeyCredential, AsyncTokenCredential]
    if args.use_token_credential:
        credential = AzureCliCredential()  # or DefaultAzureCredential() if needed
        logger.info("Using Azure token credential")
    else:
        credential = AzureKeyCredential(args.api_key)
        logger.info("Using API key credential")

    # Create and start voice assistant
    assistant = BasicVoiceAssistant(
        endpoint=args.endpoint,
        credential=credential,
        model=args.model,
        voice=args.voice,
        instructions=args.instructions,
    )

    # Setup signal handlers for graceful shutdown
    def signal_handler(_sig, _frame):
        logger.info("Received shutdown signal")
        raise KeyboardInterrupt()

    signal.signal(signal.SIGINT, signal_handler)
    signal.signal(signal.SIGTERM, signal_handler)

    # Start the assistant
    try:
        asyncio.run(assistant.start())
    except KeyboardInterrupt:
        print("\n👋 Voice assistant shut down. Goodbye!")
    except Exception as e:
        print("Fatal Error: ", e)

if __name__ == "__main__":
    # Check audio system
    try:
        p = pyaudio.PyAudio()
        # Check for input devices
        input_devices = [
            i
            for i in range(p.get_device_count())
            if cast(Union[int, float], p.get_device_info_by_index(i).get("maxInputChannels", 0) or 0) > 0
        ]
        # Check for output devices
        output_devices = [
            i
            for i in range(p.get_device_count())
            if cast(Union[int, float], p.get_device_info_by_index(i).get("maxOutputChannels", 0) or 0) > 0
        ]
        p.terminate()

        if not input_devices:
            print("❌ No audio input devices found. Please check your microphone.")
            sys.exit(1)
        if not output_devices:
            print("❌ No audio output devices found. Please check your speakers.")
            sys.exit(1)

    except Exception as e:
        print(f"❌ Audio system check failed: {e}")
        sys.exit(1)

    print("🎙️  Basic Voice Assistant with Azure VoiceLive SDK")
    print("=" * 50)

    # Run the assistant
    main()

  2. 使用以下命令登录到Azure：

    az login

  3. 运行Python文件。

    python voice-live-quickstart.py --use-token-credential

  4. Voice Live API 开始根据模型的初始响应返回音频。 可以通过说话来中断模型。 输入“Ctrl+C”退出对话。

输出

脚本的输出将打印到控制台。 你会看到指示系统状态的消息。 音频通过扬声器或耳机播放。

============================================================
🎤 VOICE ASSISTANT READY
Start speaking to begin conversation
Press Ctrl+C to exit
============================================================

🎤 Listening...
🤔 Processing...
🎤 Ready for next input...
🎤 Listening...
🤔 Processing...
🎤 Ready for next input...
🎤 Listening...
🤔 Processing...
🎤 Ready for next input...
🎤 Listening...
🤔 Processing...
🎤 Listening...
🎤 Ready for next input...
🤔 Processing...
🎤 Ready for next input...

运行的脚本将创建一个在文件夹中命名 <timestamp>_voicelive.log 的 logs 日志文件。

默认日志级别设置为INFO，但您可以通过命令行参数--verbose在启动时修改，或者通过如下所示的代码更新日志配置来更改日志级别：

logging.basicConfig(
    filename=f'logs/{timestamp}_voicelive.log',
    filemode="w",
    format='%(asctime)s:%(name)s:%(levelname)s:%(message)s',
    level=logging.INFO
)

日志文件包含有关与语音实时 API 的连接的信息，包括请求和响应数据。 可以查看日志文件以查看聊天的详细信息。

2025-10-02 14:47:37,901:__main__:INFO:Using Azure token credential
2025-10-02 14:47:37,901:__main__:INFO:Connecting to VoiceLive API with model gpt-realtime
2025-10-02 14:47:37,901:azure.core.pipeline.policies.http_logging_policy:INFO:Request URL: 'https://login.microsoftonline.com/organizations/v2.0/.well-known/openid-configuration'
Request method: 'GET'
Request headers:
    'User-Agent': 'azsdk-python-identity/1.22.0 Python/3.11.9 (Windows-10-10.0.26200-SP0)'
No body was attached to the request
2025-10-02 14:47:38,057:azure.core.pipeline.policies.http_logging_policy:INFO:Response status: 200
Response headers:
    'Date': 'Thu, 02 Oct 2025 21:47:37 GMT'
    'Content-Type': 'application/json; charset=utf-8'
    'Content-Length': '1641'
    'Connection': 'keep-alive'
    'Cache-Control': 'max-age=86400, private'
    'Strict-Transport-Security': 'REDACTED'
    'X-Content-Type-Options': 'REDACTED'
    'Access-Control-Allow-Origin': 'REDACTED'
    'Access-Control-Allow-Methods': 'REDACTED'
    'P3P': 'REDACTED'
    'x-ms-request-id': 'f81adfa1-8aa3-4ab6-a7b8-908f411e0d00'
    'x-ms-ests-server': 'REDACTED'
    'x-ms-srs': 'REDACTED'
    'Content-Security-Policy-Report-Only': 'REDACTED'
    'Cross-Origin-Opener-Policy-Report-Only': 'REDACTED'
    'Reporting-Endpoints': 'REDACTED'
    'X-XSS-Protection': 'REDACTED'
    'Set-Cookie': 'REDACTED'
    'X-Cache': 'REDACTED'
2025-10-02 14:47:42,105:azure.core.pipeline.policies.http_logging_policy:INFO:Request URL: 'https://login.microsoftonline.com/organizations/oauth2/v2.0/token'
Request method: 'POST'
Request headers:
    'Accept': 'application/json'
    'x-client-sku': 'REDACTED'
    'x-client-ver': 'REDACTED'
    'x-client-os': 'REDACTED'
    'x-ms-lib-capability': 'REDACTED'
    'client-request-id': 'REDACTED'
    'x-client-current-telemetry': 'REDACTED'
    'x-client-last-telemetry': 'REDACTED'
    'X-AnchorMailbox': 'REDACTED'
    'User-Agent': 'azsdk-python-identity/1.22.0 Python/3.11.9 (Windows-10-10.0.26200-SP0)'
A body is sent with the request
2025-10-02 14:47:42,466:azure.core.pipeline.policies.http_logging_policy:INFO:Response status: 200
Response headers:
    'Date': 'Thu, 02 Oct 2025 21:47:42 GMT'
    'Content-Type': 'application/json; charset=utf-8'
    'Content-Length': '6587'
    'Connection': 'keep-alive'
    'Cache-Control': 'no-store, no-cache'
    'Pragma': 'no-cache'
    'Expires': '-1'
    'Strict-Transport-Security': 'REDACTED'
    'X-Content-Type-Options': 'REDACTED'
    'P3P': 'REDACTED'
    'client-request-id': 'REDACTED'
    'x-ms-request-id': '2e82e728-22c0-4568-b3ed-f00ec79a2500'
    'x-ms-ests-server': 'REDACTED'
    'x-ms-clitelem': 'REDACTED'
    'x-ms-srs': 'REDACTED'
    'Content-Security-Policy-Report-Only': 'REDACTED'
    'Cross-Origin-Opener-Policy-Report-Only': 'REDACTED'
    'Reporting-Endpoints': 'REDACTED'
    'X-XSS-Protection': 'REDACTED'
    'Set-Cookie': 'REDACTED'
    'X-Cache': 'REDACTED'
2025-10-02 14:47:42,467:azure.identity._internal.interactive:INFO:InteractiveBrowserCredential.get_token succeeded
2025-10-02 14:47:42,884:__main__:INFO:AudioProcessor initialized with 24kHz PCM16 mono audio
2025-10-02 14:47:42,884:__main__:INFO:Setting up voice conversation session...
2025-10-02 14:47:42,887:__main__:INFO:Session configuration sent
2025-10-02 14:47:42,943:__main__:INFO:Audio playback system ready
2025-10-02 14:47:42,943:__main__:INFO:Voice assistant ready! Start speaking...
2025-10-02 14:47:42,975:__main__:INFO:Session ready: sess_CMLRGjWnakODcHn583fXf
2025-10-02 14:47:42,994:__main__:INFO:Started audio capture
2025-10-02 14:47:47,513:__main__:INFO:\U0001f3a4 User started speaking - stopping playback
2025-10-02 14:47:47,593:__main__:INFO:Stopped audio playback
2025-10-02 14:47:51,757:__main__:INFO:\U0001f3a4 User stopped speaking
2025-10-02 14:47:51,813:__main__:INFO:Audio playback system ready
2025-10-02 14:47:51,816:__main__:INFO:\U0001f916 Assistant response created
2025-10-02 14:47:58,009:__main__:INFO:\U0001f916 Assistant finished speaking
2025-10-02 14:47:58,009:__main__:INFO:\u2705 Response complete
2025-10-02 14:48:07,309:__main__:INFO:Received shutdown signal

本文介绍如何使用适用于 C# 的 VoiceLive SDK 将 Voice Live 与 Microsoft Foundry 模型配合使用。

参考文档 | 包（NuGet） | 在 GitHub 上的更多示例

创建并运行一个应用程序，将 Voice Live 直接与生成型 AI 模型结合使用，以创建实时语音代理。

  • 使用模型直接允许为每个会话指定自定义指令（提示），为动态或实验用例提供更大的灵活性。

  • 如果希望对会话参数进行精细控制，或者需要频繁调整提示或配置，而无需在门户中更新代理，则模型可能更可取。

  • 基于模型的会话的代码在某些方面更简单，因为它不需要管理代理 ID 或特定于代理的设置。

  • 直接模型使用适用于不需要代理级抽象或内置逻辑的方案。

若要改用语音实时 API 和代理，请参阅 语音实时 API 代理快速入门。

先决条件

启动语音对话

按照以下步骤创建控制台应用程序并安装语音 SDK。

  1. 在需要新项目的文件夹中打开命令提示符窗口。 运行以下命令，使用 .NET CLI 创建控制台应用程序。

    dotnet new console

    该命令会在你的项目目录中创建 Program.cs 文件。

  2. 使用 .NET CLI 在新项目中安装 Voice Live SDK、Azure Identity 和 NAudio。

    dotnet add package Azure.AI.VoiceLive
dotnet add package Azure.Identity
dotnet add package NAudio
dotnet add package System.CommandLine --version 2.0.0-beta4.22272.1
dotnet add package Microsoft.Extensions.Configuration.Json
dotnet add package Microsoft.Extensions.Configuration.EnvironmentVariables
dotnet add package Microsoft.Extensions.Logging.Console

  3. 在要在其中运行代码的文件夹中创建一 appsettings.json 个名为的新文件。 在该文件中，添加以下 JSON 内容：

    {
  "VoiceLive": {
    "ApiKey": "your-api-key-here",
    "Endpoint": "https://your-resource-name.services.ai.azure.com/",
    "Model": "gpt-realtime",
    "Voice": "en-US-Ava:DragonHDLatestNeural",
    "Instructions": "You are a helpful AI assistant. Respond naturally and conversationally. Keep your responses concise but engaging."
  },
  "Logging": {
    "LogLevel": {
      "Default": "Information",
      "Azure.AI.VoiceLive": "Debug"
    }
  }
}

    本快速入门中的示例代码使用Microsoft Entra ID或 API 密钥进行身份验证。 可以将脚本参数设置为 API 密钥或访问令牌。 建议使用 Microsoft Entra ID 身份验证，而无需设置 ApiKey 值及执行带有 --use-token-credential 参数的快速启动。

    将 ApiKey 值（可选）替换为 Foundry API 密钥，并将 Endpoint 值替换为资源终端节点。 还可以根据需要更改模型、语音和说明值。

    详细了解无密钥身份验证，以及如何设置环境变量。

  4. 在文件中 csharp.csproj 添加以下信息以连接 appsettings.json：

    <ItemGroup>
<None Update="appsettings.json">
    <CopyToOutputDirectory>PreserveNewest</CopyToOutputDirectory>
</None>
</ItemGroup>

  5. 将 Program.cs 的内容替换为以下代码。 此代码使用内置模型之一创建基本语音代理。 要查看更详细的版本，请参阅 GitHub 上的示例。

    // Copyright (c) Microsoft Corporation. All rights reserved.
// Licensed under the MIT License.

using System;
using System.CommandLine;
using System.Threading;
using System.Threading.Tasks;
using System.Threading.Channels;
using System.Collections.Generic;
using Azure.AI.VoiceLive;
using Azure.Core;
using Azure.Core.Pipeline;
using Azure.Identity;
using Microsoft.Extensions.Configuration;
using Microsoft.Extensions.Logging;
using NAudio.Wave;

namespace Azure.AI.VoiceLive.Samples
{
    /// <summary>
    /// FILE: Program.cs (Consolidated)
    /// </summary>
    /// <remarks>
    /// DESCRIPTION:
    ///     This consolidated sample demonstrates the fundamental capabilities of the VoiceLive SDK by creating
    ///     a basic voice assistant that can engage in natural conversation with proper interruption
    ///     handling. This serves as the foundational example that showcases the core value
    ///     proposition of unified speech-to-speech interaction.
    ///     
    ///     All necessary code has been consolidated into this single file for easy distribution and execution.
    ///
    /// USAGE:
    ///     dotnet run
    ///
    ///     Set the environment variables with your own values before running the sample:
    ///     1) AZURE_VOICELIVE_API_KEY - The Azure VoiceLive API key
    ///     2) AZURE_VOICELIVE_ENDPOINT - The Azure VoiceLive endpoint
    ///
    ///     Or update appsettings.json with your values.
    ///
    /// REQUIREMENTS:
    ///     - Azure.AI.VoiceLive
    ///     - Azure.Identity
    ///     - NAudio (for audio capture and playback)
    ///     - Microsoft.Extensions.Configuration
    ///     - System.CommandLine
    ///     - System.Threading.Channels
    /// </remarks>
    public class Program
    {
        /// <summary>
        /// Main entry point for the Voice Assistant sample.
        /// </summary>
        /// <param name="args"></param>
        /// <returns></returns>
        public static async Task<int> Main(string[] args)
        {
            // Create command line interface
            var rootCommand = CreateRootCommand();
            return await rootCommand.InvokeAsync(args).ConfigureAwait(false);
        }

        private static RootCommand CreateRootCommand()
        {
            var rootCommand = new RootCommand("Basic Voice Assistant using Azure VoiceLive SDK");

            var apiKeyOption = new Option<string?>(
                "--api-key",
                "Azure VoiceLive API key. If not provided, will use AZURE_VOICELIVE_API_KEY environment variable.");

            var endpointOption = new Option<string>(
                "--endpoint",
                () => "wss://api.voicelive.com/v1",
                "Azure VoiceLive endpoint");

            var modelOption = new Option<string>(
                "--model",
                () => "gpt-4o",
                "VoiceLive model to use");

            var voiceOption = new Option<string>(
                "--voice",
                () => "en-US-AvaNeural",
                "Voice to use for the assistant");

            var instructionsOption = new Option<string>(
                "--instructions",
                () => "You are a helpful AI assistant. Respond naturally and conversationally. Keep your responses concise but engaging.",
                "System instructions for the AI assistant");

            var useTokenCredentialOption = new Option<bool>(
                "--use-token-credential",
                "Use Azure token credential instead of API key");

            var verboseOption = new Option<bool>(
                "--verbose",
                "Enable verbose logging");

            rootCommand.AddOption(apiKeyOption);
            rootCommand.AddOption(endpointOption);
            rootCommand.AddOption(modelOption);
            rootCommand.AddOption(voiceOption);
            rootCommand.AddOption(instructionsOption);
            rootCommand.AddOption(useTokenCredentialOption);
            rootCommand.AddOption(verboseOption);

            rootCommand.SetHandler(async (
                string? apiKey,
                string endpoint,
                string model,
                string voice,
                string instructions,
                bool useTokenCredential,
                bool verbose) =>
            {
                await RunVoiceAssistantAsync(apiKey, endpoint, model, voice, instructions, useTokenCredential, verbose).ConfigureAwait(false);
            },
            apiKeyOption,
            endpointOption,
            modelOption,
            voiceOption,
            instructionsOption,
            useTokenCredentialOption,
            verboseOption);

            return rootCommand;
        }

        private static async Task RunVoiceAssistantAsync(
            string? apiKey,
            string endpoint,
            string model,
            string voice,
            string instructions,
            bool useTokenCredential,
            bool verbose)
        {
            // Setup configuration
            var configuration = new ConfigurationBuilder()
                .AddJsonFile("appsettings.json", optional: true)
                .AddEnvironmentVariables()
                .Build();

            // Override with command line values if provided
            apiKey ??= configuration["VoiceLive:ApiKey"] ?? Environment.GetEnvironmentVariable("AZURE_VOICELIVE_API_KEY");
            endpoint = configuration["VoiceLive:Endpoint"] ?? endpoint;
            model = configuration["VoiceLive:Model"] ?? model;
            voice = configuration["VoiceLive:Voice"] ?? voice;
            instructions = configuration["VoiceLive:Instructions"] ?? instructions;

            // Setup logging
            using var loggerFactory = LoggerFactory.Create(builder =>
            {
                builder.AddConsole();
                if (verbose)
                {
                    builder.SetMinimumLevel(LogLevel.Debug);
                }
                else
                {
                    builder.SetMinimumLevel(LogLevel.Information);
                }
            });

            var logger = loggerFactory.CreateLogger<Program>();

            // Validate credentials
            if (string.IsNullOrEmpty(apiKey) && !useTokenCredential)
            {
                Console.WriteLine("❌ Error: No authentication provided");
                Console.WriteLine("Please provide an API key using --api-key or set AZURE_VOICELIVE_API_KEY environment variable,");
                Console.WriteLine("or use --use-token-credential for Azure authentication.");
                return;
            }

            // Check audio system before starting
            if (!CheckAudioSystem(logger))
            {
                return;
            }

            try
            {
                // Create client with appropriate credential
                VoiceLiveClient client;
                if (useTokenCredential)
                {
                    var tokenCredential = new DefaultAzureCredential();
                    client = new VoiceLiveClient(new Uri(endpoint), tokenCredential, new VoiceLiveClientOptions());
                    logger.LogInformation("Using Azure token credential");
                }
                else
                {
                    var keyCredential = new Azure.AzureKeyCredential(apiKey!);
                    client = new VoiceLiveClient(new Uri(endpoint), keyCredential, new VoiceLiveClientOptions());
                    logger.LogInformation("Using API key credential");
                }

                // Create and start voice assistant
                using var assistant = new BasicVoiceAssistant(
                    client,
                    model,
                    voice,
                    instructions,
                    loggerFactory);

                // Setup cancellation token for graceful shutdown
                using var cancellationTokenSource = new CancellationTokenSource();
                Console.CancelKeyPress += (sender, e) =>
                {
                    e.Cancel = true;
                    logger.LogInformation("Received shutdown signal");
                    cancellationTokenSource.Cancel();
                };

                // Start the assistant
                await assistant.StartAsync(cancellationTokenSource.Token).ConfigureAwait(false);
            }
            catch (OperationCanceledException)
            {
                Console.WriteLine("\n👋 Voice assistant shut down. Goodbye!");
            }
            catch (Exception ex)
            {
                logger.LogError(ex, "Fatal error");
                Console.WriteLine($"❌ Error: {ex.Message}");
            }
        }

        private static bool CheckAudioSystem(ILogger logger)
        {
            try
            {
                // Try input (default device)
                using (var waveIn = new WaveInEvent
                {
                    WaveFormat = new WaveFormat(24000, 16, 1),
                    BufferMilliseconds = 50
                })
                {
                    // Start/Stop to force initialization and surface any device errors
                    waveIn.DataAvailable += (_, __) => { };
                    waveIn.StartRecording();
                    waveIn.StopRecording();
                }

                // Try output (default device)
                var buffer = new BufferedWaveProvider(new WaveFormat(24000, 16, 1))
                {
                    BufferDuration = TimeSpan.FromMilliseconds(200)
                };

                using (var waveOut = new WaveOutEvent { DesiredLatency = 100 })
                {
                    waveOut.Init(buffer);
                    // Playing isn't strictly required to validate a device, but it's safe
                    waveOut.Play();
                    waveOut.Stop();
                }

                logger.LogInformation("Audio system check passed (default input/output initialized).");
                return true;
            }
            catch (Exception ex)
            {
                Console.WriteLine($"❌ Audio system check failed: {ex.Message}");
                return false;
            }
        }
    }

    /// <summary>
    /// Basic voice assistant implementing the VoiceLive SDK patterns.
    ///</summary>
    /// <remarks>
    /// This sample now demonstrates some of the new convenience methods added to the VoiceLive SDK:
    /// - ClearStreamingAudioAsync() - Clears all input audio currently being streamed
    /// - CancelResponseAsync() - Cancels the current response generation (existing method)
    /// - ConfigureSessionAsync() - Configures session options (existing method)
    ///
    /// Additional convenience methods available but not shown in this sample:
    /// - StartAudioTurnAsync() / EndAudioTurnAsync() / CancelAudioTurnAsync() - Audio turn management
    /// - AppendAudioToTurnAsync() - Append audio data to an ongoing turn
    /// - ConnectAvatarAsync() - Connect avatar with SDP for media negotiation
    ///
    /// These methods provide a more developer-friendly API similar to the OpenAI SDK,
    /// eliminating the need to manually construct and populate ClientEvent classes.
    /// </remarks>
    public class BasicVoiceAssistant : IDisposable
    {
        private readonly VoiceLiveClient _client;
        private readonly string _model;
        private readonly string _voice;
        private readonly string _instructions;
        private readonly ILogger<BasicVoiceAssistant> _logger;
        private readonly ILoggerFactory _loggerFactory;

        private VoiceLiveSession? _session;
        private AudioProcessor? _audioProcessor;
        private bool _disposed;
    // Tracks whether an assistant response is currently active (created and not yet completed)
    private bool _responseActive;
    // Tracks whether the assistant can still cancel the current response (between ResponseCreated and ResponseDone)
    private bool _canCancelResponse;

        /// <summary>
        /// Initializes a new instance of the BasicVoiceAssistant class.
        /// </summary>
        /// <param name="client">The VoiceLive client.</param>
        /// <param name="model">The model to use.</param>
        /// <param name="voice">The voice to use.</param>
        /// <param name="instructions">The system instructions.</param>
        /// <param name="loggerFactory">Logger factory for creating loggers.</param>
        public BasicVoiceAssistant(
            VoiceLiveClient client,
            string model,
            string voice,
            string instructions,
            ILoggerFactory loggerFactory)
        {
            _client = client ?? throw new ArgumentNullException(nameof(client));
            _model = model ?? throw new ArgumentNullException(nameof(model));
            _voice = voice ?? throw new ArgumentNullException(nameof(voice));
            _instructions = instructions ?? throw new ArgumentNullException(nameof(instructions));
            _loggerFactory = loggerFactory ?? throw new ArgumentNullException(nameof(loggerFactory));
            _logger = loggerFactory.CreateLogger<BasicVoiceAssistant>();
        }

        /// <summary>
        /// Start the voice assistant session.
        /// </summary>
        /// <param name="cancellationToken">Cancellation token for stopping the session.</param>
        public async Task StartAsync(CancellationToken cancellationToken = default)
        {
            try
            {
                _logger.LogInformation("Connecting to VoiceLive API with model {Model}", _model);

                // Start VoiceLive session
                _session = await _client.StartSessionAsync(_model, cancellationToken).ConfigureAwait(false);

                // Initialize audio processor
                _audioProcessor = new AudioProcessor(_session, _loggerFactory.CreateLogger<AudioProcessor>());

                // Configure session for voice conversation
                await SetupSessionAsync(cancellationToken).ConfigureAwait(false);

                // Start audio systems
                await _audioProcessor.StartPlaybackAsync().ConfigureAwait(false);
                await _audioProcessor.StartCaptureAsync().ConfigureAwait(false);

                _logger.LogInformation("Voice assistant ready! Start speaking...");
                Console.WriteLine();
                Console.WriteLine("=" + new string('=', 59));
                Console.WriteLine("🎤 VOICE ASSISTANT READY");
                Console.WriteLine("Start speaking to begin conversation");
                Console.WriteLine("Press Ctrl+C to exit");
                Console.WriteLine("=" + new string('=', 59));
                Console.WriteLine();

                // Process events
                await ProcessEventsAsync(cancellationToken).ConfigureAwait(false);
            }
            catch (OperationCanceledException)
            {
                _logger.LogInformation("Received cancellation signal, shutting down...");
            }
            catch (Exception ex)
            {
                _logger.LogError(ex, "Connection error");
                throw;
            }
            finally
            {
                // Cleanup
                if (_audioProcessor != null)
                {
                    await _audioProcessor.CleanupAsync().ConfigureAwait(false);
                }
            }
        }

        /// <summary>
        /// Configure the VoiceLive session for audio conversation.
        /// </summary>
        private async Task SetupSessionAsync(CancellationToken cancellationToken)
        {
            _logger.LogInformation("Setting up voice conversation session...");

            // Azure voice
            var azureVoice = new AzureStandardVoice(_voice);

            // Create strongly typed turn detection configuration
            var turnDetectionConfig = new ServerVadTurnDetection
            {
                Threshold = 0.5f,
                PrefixPadding = TimeSpan.FromMilliseconds(300),
                SilenceDuration = TimeSpan.FromMilliseconds(500)
            };

            // Create conversation session options
            var sessionOptions = new VoiceLiveSessionOptions
            {
                InputAudioEchoCancellation = new AudioEchoCancellation(),
                Model = _model,
                Instructions = _instructions,
                Voice = azureVoice,
                InputAudioFormat = InputAudioFormat.Pcm16,
                OutputAudioFormat = OutputAudioFormat.Pcm16,
                TurnDetection = turnDetectionConfig
            };

            // Ensure modalities include audio
            sessionOptions.Modalities.Clear();
            sessionOptions.Modalities.Add(InteractionModality.Text);
            sessionOptions.Modalities.Add(InteractionModality.Audio);

            await _session!.ConfigureSessionAsync(sessionOptions, cancellationToken).ConfigureAwait(false);

            _logger.LogInformation("Session configuration sent");
        }

        /// <summary>
        /// Process events from the VoiceLive session.
        /// </summary>
        private async Task ProcessEventsAsync(CancellationToken cancellationToken)
        {
            try
            {
                await foreach (SessionUpdate serverEvent in _session!.GetUpdatesAsync(cancellationToken).ConfigureAwait(false))
                {
                    await HandleSessionUpdateAsync(serverEvent, cancellationToken).ConfigureAwait(false);
                }
            }
            catch (OperationCanceledException)
            {
                _logger.LogInformation("Event processing cancelled");
            }
            catch (Exception ex)
            {
                _logger.LogError(ex, "Error processing events");
                throw;
            }
        }

        /// <summary>
        /// Handle different types of server events from VoiceLive.
        /// </summary>
        private async Task HandleSessionUpdateAsync(SessionUpdate serverEvent, CancellationToken cancellationToken)
        {
            _logger.LogDebug("Received event: {EventType}", serverEvent.GetType().Name);

            switch (serverEvent)
            {
                case SessionUpdateSessionCreated sessionCreated:
                    await HandleSessionCreatedAsync(sessionCreated, cancellationToken).ConfigureAwait(false);
                    break;

                case SessionUpdateSessionUpdated sessionUpdated:
                    _logger.LogInformation("Session updated successfully");

                    // Start audio capture once session is ready
                    if (_audioProcessor != null)
                    {
                        await _audioProcessor.StartCaptureAsync().ConfigureAwait(false);
                    }
                    break;

                case SessionUpdateInputAudioBufferSpeechStarted speechStarted:
                    _logger.LogInformation("🎤 User started speaking - stopping playback");
                    Console.WriteLine("🎤 Listening...");

                    // Stop current assistant audio playback (interruption handling)
                    if (_audioProcessor != null)
                    {
                        await _audioProcessor.StopPlaybackAsync().ConfigureAwait(false);
                    }

                    // Only attempt cancellation / clearing if a response is active and cancellable
                    if (_responseActive && _canCancelResponse)
                    {
                        // Cancel any ongoing response
                        try
                        {
                            await _session!.CancelResponseAsync(cancellationToken).ConfigureAwait(false);
                            _logger.LogInformation("🛑 Active response cancelled due to user barge-in");
                        }
                        catch (Exception ex)
                        {
                            if (ex.Message.Contains("no active response", StringComparison.OrdinalIgnoreCase))
                            {
                                _logger.LogDebug("Cancellation benign: response already completed");
                            }
                            else
                            {
                                _logger.LogWarning(ex, "Response cancellation failed during barge-in");
                            }
                        }

                        // Clear any streaming audio still in transit
                        try
                        {
                            await _session!.ClearStreamingAudioAsync(cancellationToken).ConfigureAwait(false);
                            _logger.LogInformation("✨ Cleared streaming audio after cancellation");
                        }
                        catch (Exception ex)
                        {
                            _logger.LogDebug(ex, "ClearStreamingAudio call failed (may not be supported in all scenarios)");
                        }
                    }
                    else
                    {
                        _logger.LogDebug("No active/cancellable response during barge-in; skipping cancellation");
                    }
                    break;

                case SessionUpdateInputAudioBufferSpeechStopped speechStopped:
                    _logger.LogInformation("🎤 User stopped speaking");
                    Console.WriteLine("🤔 Processing...");

                    // Restart playback system for response
                    if (_audioProcessor != null)
                    {
                        await _audioProcessor.StartPlaybackAsync().ConfigureAwait(false);
                    }
                    break;

                case SessionUpdateResponseCreated responseCreated:
                    _logger.LogInformation("🤖 Assistant response created");
                    _responseActive = true;
                    _canCancelResponse = true;
                    break;

                case SessionUpdateResponseAudioDelta audioDelta:
                    // Stream audio response to speakers
                    _logger.LogDebug("Received audio delta");

                    if (audioDelta.Delta != null && _audioProcessor != null)
                    {
                        byte[] audioData = audioDelta.Delta.ToArray();
                        await _audioProcessor.QueueAudioAsync(audioData).ConfigureAwait(false);
                    }
                    break;

                case SessionUpdateResponseAudioDone audioDone:
                    _logger.LogInformation("🤖 Assistant finished speaking");
                    Console.WriteLine("🎤 Ready for next input...");
                    break;

                case SessionUpdateResponseDone responseDone:
                    _logger.LogInformation("✅ Response complete");
                    _responseActive = false;
                    _canCancelResponse = false;
                    break;

                case SessionUpdateError errorEvent:
                    _logger.LogError("❌ VoiceLive error: {ErrorMessage}", errorEvent.Error?.Message);
                    Console.WriteLine($"Error: {errorEvent.Error?.Message}");
                    _responseActive = false;
                    _canCancelResponse = false;
                    break;

                default:
                    _logger.LogDebug("Unhandled event type: {EventType}", serverEvent.GetType().Name);
                    break;
            }
        }

        /// <summary>
        /// Handle session created event.
        /// </summary>
        private async Task HandleSessionCreatedAsync(SessionUpdateSessionCreated sessionCreated, CancellationToken cancellationToken)
        {
            _logger.LogInformation("Session ready: {SessionId}", sessionCreated.Session?.Id);

            // Start audio capture once session is ready
            if (_audioProcessor != null)
            {
                await _audioProcessor.StartCaptureAsync().ConfigureAwait(false);
            }
        }

        /// <summary>
        /// Dispose of resources.
        /// </summary>
        public void Dispose()
        {
            if (_disposed)
                return;

            _audioProcessor?.Dispose();
            _session?.Dispose();
            _disposed = true;
        }
    }

    /// <summary>
    /// Handles real-time audio capture and playback for the voice assistant.
    ///
    /// This processor demonstrates some of the new VoiceLive SDK convenience methods:
    /// - Uses existing SendInputAudioAsync() method for audio streaming
    /// - Shows how convenience methods simplify audio operations
    ///
    /// Additional convenience methods available in the SDK:
    /// - StartAudioTurnAsync() / AppendAudioToTurnAsync() / EndAudioTurnAsync() - Audio turn management
    /// - ClearStreamingAudioAsync() - Clear all streaming audio
    /// - ConnectAvatarAsync() - Avatar connection with SDP
    ///
    /// Threading Architecture:
    /// - Main thread: Event loop and UI
    /// - Capture thread: NAudio input stream reading
    /// - Send thread: Async audio data transmission to VoiceLive
    /// - Playback thread: NAudio output stream writing
    /// </summary>
    public class AudioProcessor : IDisposable
    {
        private readonly VoiceLiveSession _session;
        private readonly ILogger<AudioProcessor> _logger;

        // Audio configuration - PCM16, 24kHz, mono as specified
        private const int SampleRate = 24000;
        private const int Channels = 1;
        private const int BitsPerSample = 16;

        // NAudio components
        private WaveInEvent? _waveIn;
        private WaveOutEvent? _waveOut;
        private BufferedWaveProvider? _playbackBuffer;

        // Audio capture and playback state
        private bool _isCapturing;
        private bool _isPlaying;

        // Audio streaming channels
        private readonly Channel<byte[]> _audioSendChannel;
        private readonly Channel<byte[]> _audioPlaybackChannel;
        private readonly ChannelWriter<byte[]> _audioSendWriter;
        private readonly ChannelReader<byte[]> _audioSendReader;
        private readonly ChannelWriter<byte[]> _audioPlaybackWriter;
        private readonly ChannelReader<byte[]> _audioPlaybackReader;

        // Background tasks
        private Task? _audioSendTask;
        private Task? _audioPlaybackTask;
        private readonly CancellationTokenSource _cancellationTokenSource;
        private CancellationTokenSource _playbackCancellationTokenSource;

        /// <summary>
        /// Initializes a new instance of the AudioProcessor class.
        /// </summary>
        /// <param name="session">The VoiceLive session for audio communication.</param>
        /// <param name="logger">Logger for diagnostic information.</param>
        public AudioProcessor(VoiceLiveSession session, ILogger<AudioProcessor> logger)
        {
            _session = session ?? throw new ArgumentNullException(nameof(session));
            _logger = logger ?? throw new ArgumentNullException(nameof(logger));

            // Create unbounded channels for audio data
            _audioSendChannel = Channel.CreateUnbounded<byte[]>();
            _audioSendWriter = _audioSendChannel.Writer;
            _audioSendReader = _audioSendChannel.Reader;

            _audioPlaybackChannel = Channel.CreateUnbounded<byte[]>();
            _audioPlaybackWriter = _audioPlaybackChannel.Writer;
            _audioPlaybackReader = _audioPlaybackChannel.Reader;

            _cancellationTokenSource = new CancellationTokenSource();
            _playbackCancellationTokenSource = new CancellationTokenSource();

            _logger.LogInformation("AudioProcessor initialized with {SampleRate}Hz PCM16 mono audio", SampleRate);
        }

        /// <summary>
        /// Start capturing audio from microphone.
        /// </summary>
        public Task StartCaptureAsync()
        {
            if (_isCapturing)
                return Task.CompletedTask;

            _isCapturing = true;

            try
            {
                _waveIn = new WaveInEvent
                {
                    WaveFormat = new WaveFormat(SampleRate, BitsPerSample, Channels),
                    BufferMilliseconds = 50 // 50ms buffer for low latency
                };

                _waveIn.DataAvailable += OnAudioDataAvailable;
                _waveIn.RecordingStopped += OnRecordingStopped;

                _waveIn.DeviceNumber = 0;

                _waveIn.StartRecording();

                // Start audio send task
                _audioSendTask = ProcessAudioSendAsync(_cancellationTokenSource.Token);

                _logger.LogInformation("Started audio capture");
                return Task.CompletedTask;
            }
            catch (Exception ex)
            {
                _logger.LogError(ex, "Failed to start audio capture");
                _isCapturing = false;
                throw;
            }
        }

        /// <summary>
        /// Stop capturing audio.
        /// </summary>
        public async Task StopCaptureAsync()
        {
            if (!_isCapturing)
                return;

            _isCapturing = false;

            if (_waveIn != null)
            {
                _waveIn.StopRecording();
                _waveIn.DataAvailable -= OnAudioDataAvailable;
                _waveIn.RecordingStopped -= OnRecordingStopped;
                _waveIn.Dispose();
                _waveIn = null;
            }

            // Complete the send channel and wait for the send task
            _audioSendWriter.TryComplete();
            if (_audioSendTask != null)
            {
                await _audioSendTask.ConfigureAwait(false);
                _audioSendTask = null;
            }

            _logger.LogInformation("Stopped audio capture");
        }

        /// <summary>
        /// Initialize audio playback system.
        /// </summary>
        public Task StartPlaybackAsync()
        {
            if (_isPlaying)
                return Task.CompletedTask;

            _isPlaying = true;

            try
            {
                _waveOut = new WaveOutEvent
                {
                    DesiredLatency = 100 // 100ms latency
                };

                _playbackBuffer = new BufferedWaveProvider(new WaveFormat(SampleRate, BitsPerSample, Channels))
                {
                    BufferDuration = TimeSpan.FromSeconds(10), // 10 second buffer
                    DiscardOnBufferOverflow = true
                };

                _waveOut.Init(_playbackBuffer);
                _waveOut.Play();

                _playbackCancellationTokenSource = new CancellationTokenSource();

                // Start audio playback task
                _audioPlaybackTask = ProcessAudioPlaybackAsync();

                _logger.LogInformation("Audio playback system ready");
                return Task.CompletedTask;
            }
            catch (Exception ex)
            {
                _logger.LogError(ex, "Failed to initialize audio playback");
                _isPlaying = false;
                throw;
            }
        }

        /// <summary>
        /// Stop audio playback and clear buffer.
        /// </summary>
        public async Task StopPlaybackAsync()
        {
            if (!_isPlaying)
                return;

            _isPlaying = false;

            // Clear the playback channel
            while (_audioPlaybackReader.TryRead(out _))
            { }

            if (_playbackBuffer != null)
            {
                _playbackBuffer.ClearBuffer();
            }

            if (_waveOut != null)
            {
                _waveOut.Stop();
                _waveOut.Dispose();
                _waveOut = null;
            }

            _playbackBuffer = null;

            // Complete the playback channel and wait for the playback task
            _playbackCancellationTokenSource.Cancel();

            if (_audioPlaybackTask != null)
            {
                await _audioPlaybackTask.ConfigureAwait(false);
                _audioPlaybackTask = null;
            }

            _logger.LogInformation("Stopped audio playback");
        }

        /// <summary>
        /// Queue audio data for playback.
        /// </summary>
        /// <param name="audioData">The audio data to queue.</param>
        public async Task QueueAudioAsync(byte[] audioData)
        {
            if (_isPlaying && audioData.Length > 0)
            {
                await _audioPlaybackWriter.WriteAsync(audioData).ConfigureAwait(false);
            }
        }

        /// <summary>
        /// Event handler for audio data available from microphone.
        /// </summary>
        private void OnAudioDataAvailable(object? sender, WaveInEventArgs e)
        {
            if (_isCapturing && e.BytesRecorded > 0)
            {
                byte[] audioData = new byte[e.BytesRecorded];
                Array.Copy(e.Buffer, 0, audioData, 0, e.BytesRecorded);

                // Queue audio data for sending (non-blocking)
                if (!_audioSendWriter.TryWrite(audioData))
                {
                    _logger.LogWarning("Failed to queue audio data for sending - channel may be full");
                }
            }
        }

        /// <summary>
        /// Event handler for recording stopped.
        /// </summary>
        private void OnRecordingStopped(object? sender, StoppedEventArgs e)
        {
            if (e.Exception != null)
            {
                _logger.LogError(e.Exception, "Audio recording stopped due to error");
            }
        }

        /// <summary>
        /// Background task to process audio data and send to VoiceLive service.
        /// </summary>
        private async Task ProcessAudioSendAsync(CancellationToken cancellationToken)
        {
            try
            {
                await foreach (byte[] audioData in _audioSendReader.ReadAllAsync(cancellationToken).ConfigureAwait(false))
                {
                    if (cancellationToken.IsCancellationRequested)
                        break;

                    try
                    {
                        // Send audio data directly to the session using the convenience method
                        // This demonstrates the existing SendInputAudioAsync convenience method
                        // Other available methods: StartAudioTurnAsync, AppendAudioToTurnAsync, EndAudioTurnAsync
                        await _session.SendInputAudioAsync(audioData, cancellationToken).ConfigureAwait(false);
                    }
                    catch (Exception ex)
                    {
                        _logger.LogError(ex, "Error sending audio data to VoiceLive");
                        // Continue processing other audio data
                    }
                }
            }
            catch (OperationCanceledException)
            {
                // Expected when cancellation is requested
            }
            catch (Exception ex)
            {
                _logger.LogError(ex, "Error in audio send processing");
            }
        }

        /// <summary>
        /// Background task to process audio playback.
        /// </summary>
        private async Task ProcessAudioPlaybackAsync()
        {
            try
            {
                CancellationTokenSource combinedTokenSource = CancellationTokenSource.CreateLinkedTokenSource(_playbackCancellationTokenSource.Token, _cancellationTokenSource.Token);
                var cancellationToken = combinedTokenSource.Token;

                await foreach (byte[] audioData in _audioPlaybackReader.ReadAllAsync(cancellationToken).ConfigureAwait(false))
                {
                    if (cancellationToken.IsCancellationRequested)
                        break;

                    try
                    {
                        if (_playbackBuffer != null && _isPlaying)
                        {
                            _playbackBuffer.AddSamples(audioData, 0, audioData.Length);
                        }
                    }
                    catch (Exception ex)
                    {
                        _logger.LogError(ex, "Error in audio playback");
                        // Continue processing other audio data
                    }
                }
            }
            catch (OperationCanceledException)
            {
                // Expected when cancellation is requested
            }
            catch (Exception ex)
            {
                _logger.LogError(ex, "Error in audio playback processing");
            }
        }

        /// <summary>
        /// Clean up audio resources.
        /// </summary>
        public async Task CleanupAsync()
        {
            await StopCaptureAsync().ConfigureAwait(false);
            await StopPlaybackAsync().ConfigureAwait(false);

            _cancellationTokenSource.Cancel();

            // Wait for background tasks to complete
            var tasks = new List<Task>();
            if (_audioSendTask != null)
                tasks.Add(_audioSendTask);
            if (_audioPlaybackTask != null)
                tasks.Add(_audioPlaybackTask);

            if (tasks.Count > 0)
            {
                await Task.WhenAll(tasks).ConfigureAwait(false);
            }

            _logger.LogInformation("Audio processor cleaned up");
        }

        /// <summary>
        /// Dispose of resources.
        /// </summary>
        public void Dispose()
        {
            CleanupAsync().Wait();
            _cancellationTokenSource.Dispose();
        }
    }
}

  6. 运行控制台应用程序以启动实时对话：

    dotnet run --use-token-credential

输出

脚本的输出将打印到控制台。 你会看到指示连接状态、音频流和播放的消息。 音频通过扬声器或耳机播放。

info: Azure.AI.VoiceLive.Samples.Program[0]
      Audio system check passed (default input/output initialized).
info: Azure.AI.VoiceLive.Samples.Program[0]
      Using Azure token credential
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      Connecting to VoiceLive API with model gpt-realtime
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      AudioProcessor initialized with 24000Hz PCM16 mono audio
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      Setting up voice conversation session...
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      Session configuration sent
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Audio playback system ready
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Started audio capture
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      Voice assistant ready! Start speaking...

============================================================
🎤 VOICE ASSISTANT READY
Start speaking to begin conversation
Press Ctrl+C to exit
============================================================

info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      Session ready: sess_CVnpwfxxxxxACIzrrr7
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      Session updated successfully
🎤 Listening...
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      🎤 User started speaking - stopping playback
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Stopped audio playback
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      ✨ Used ClearStreamingAudioAsync convenience method
🤔 Processing...
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      🎤 User stopped speaking
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Audio playback system ready
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      🤖 Assistant response created
🎤 Ready for next input...
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      🤖 Assistant finished speaking
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      ✅ Response complete
info: Azure.AI.VoiceLive.Samples.Program[0]
      Received shutdown signal
info: Azure.AI.VoiceLive.Samples.BasicVoiceAssistant[0]
      Event processing cancelled
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Stopped audio capture
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Stopped audio playback
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Audio processor cleaned up
info: Azure.AI.VoiceLive.Samples.AudioProcessor[0]
      Audio processor cleaned up

本文介绍如何使用适用于 JavaScript 的 VoiceLive SDK 通过 Microsoft Foundry 模型 使用 Voice Live。

参考文档 | 包（npm） | GitHub上的更多示例

创建并运行一个应用程序，将 Voice Live 直接与生成型 AI 模型结合使用，以创建实时语音代理。

  • 使用模型直接允许为每个会话指定自定义指令（提示），为动态或实验用例提供更大的灵活性。

  • 如果希望对会话参数进行精细控制，或者需要频繁调整提示或配置，而无需在门户中更新代理，则模型可能更可取。

  • 基于模型的会话的代码在某些方面更简单，因为它不需要管理代理 ID 或特定于代理的设置。

  • 直接模型使用适用于不需要代理级抽象或内置逻辑的方案。

若要改用语音实时 API 和代理，请参阅 语音实时 API 代理快速入门。

注释

JavaScript 语音实时 SDK 专为具有内置 WebSocket 和 Web 音频支持的基于浏览器的应用程序而设计。 本快速入门使用 Node.js 结合node-record-lpcm16speaker，提供命令行体验。

先决条件

小窍门

若要使用 Voice Live，无需使用 Microsoft Foundry 资源部署音频模型。 Voice Live 是全托管的，模型会自动为你部署。 有关模型可用性的详细信息，请参阅 Voice Live 概述文档。

Microsoft Entra ID先决条件

若要使用 Microsoft Entra ID 进行推荐的无密钥身份验证，需要：

  • 安装用于无密钥身份验证的 Azure CLI 和 Microsoft Entra ID。
  • 将Cognitive Services User角色分配给用户帐户。 可以在 Azure 门户中的 访问控制（IAM）>添加角色分配下分配角色。

设置

  1. 创建新文件夹 voice-live-quickstart，并使用以下命令转到快速入门文件夹：

    mkdir voice-live-quickstart && cd voice-live-quickstart

  2. 创建包含以下内容的 package.json 文件：

    {
  "name": "voice-live-quickstart",
  "version": "1.0.0",
  "private": true,
  "type": "module",
  "dependencies": {
    "@azure/ai-voicelive": "1.0.0-beta.3",
    "@azure/ai-agents": "1.2.0-beta.2",
    "@azure/identity": "^4.6.0",
    "dotenv": "^16.4.7",
    "node-record-lpcm16": "^1.0.1",
    "speaker": "^0.5.5"
  }
}

  3. 安装依赖项：

    npm install

检索资源信息

在要在其中运行代码的文件夹中创建一 .env 个名为的新文件。

在 .env 文件中，添加以下用于身份验证的环境变量：

AZURE_VOICELIVE_ENDPOINT=<your_endpoint>
AZURE_VOICELIVE_MODEL=<your_model>
AZURE_VOICELIVE_API_VERSION=2025-10-01
AZURE_VOICELIVE_API_KEY=<your_api_key> # Only required if using API key authentication

将默认值替换为实际终结点、模型、API 版本和 API 密钥。

变量名称 价值
AZURE_VOICELIVE_ENDPOINT 从 Azure 门户检查资源时，可以在 Keys 和 Endpoint 节中找到此值。
AZURE_VOICELIVE_MODEL 要使用的模型。 例如，gpt-4o 或 gpt-realtime-mini。 有关模型可用性的详细信息，请参阅 Voice Live API 概述文档。
AZURE_VOICELIVE_API_VERSION 要使用的 API 版本。 例如，2025-10-01。

详细了解无密钥身份验证，以及如何设置环境变量。

启动会话

本快速入门中的示例代码支持Microsoft Entra ID和 API 密钥身份验证。 如果您首选无密钥身份验证，--use-token-credential 标志将切换到 DefaultAzureCredential。

该示例连接到 Voice Live，并使用以下字段配置会话：

  • model：要调用的模型部署名称（例如， gpt-realtime）。
  • voice：用于音频响应的语音。 支持Azure标准语音和 OpenAI 语音。
  • modalities：会话支持的模式。 使用 ["text", "audio"] 进行语音对话。
  • instructions：配置助手行为的系统级说明。
  • inputAudioFormat：输入音频流的格式。 ** 使用 pcm16 处理原始 PCM 音频。

  1. 使用以下代码创建 model-quickstart.js 文件：

    // Copyright (c) Microsoft Corporation. All rights reserved.
// Licensed under the MIT License.

import "dotenv/config";
import { VoiceLiveClient } from "@azure/ai-voicelive";
import { AzureKeyCredential } from "@azure/core-auth";
import { DefaultAzureCredential } from "@azure/identity";
import { spawn } from "node:child_process";
import { existsSync, mkdirSync, appendFileSync } from "node:fs";
import { join, dirname } from "node:path";
import { fileURLToPath } from "node:url";

const __dirname = dirname(fileURLToPath(import.meta.url));

const logsDir = join(__dirname, "logs");
if (!existsSync(logsDir)) mkdirSync(logsDir, { recursive: true });

const timestamp = new Date()
  .toISOString()
  .replace(/[:.]/g, "-")
  .replace("T", "_")
  .slice(0, 19);
const conversationLogFile = join(logsDir, `conversation_${timestamp}.log`);

function writeConversationLog(message) {
  appendFileSync(conversationLogFile, message + "\n", "utf-8");
}

function printUsage() {
  console.log("Usage: node model-quickstart.js [options]");
  console.log("");
  console.log("Options:");
  console.log("  --api-key <key>             VoiceLive API key");
  console.log("  --endpoint <url>            VoiceLive endpoint URL");
  console.log("  --model <name>              Model to use (default: gpt-realtime)");
  console.log(
    "  --voice <name>              Voice (default: en-US-Ava:DragonHDLatestNeural)",
  );
  console.log("  --instructions <text>       System instructions for the assistant");
  console.log("  --audio-input-device <name> Explicit SoX input device name (Windows)");
  console.log("  --list-audio-devices        List available audio input devices and exit");
  console.log("  --greeting-text <text>      Send a pre-defined greeting instead of LLM-generated");
  console.log("  --use-token-credential      Use Azure credential instead of API key");
  console.log("  --no-audio                  Connect and configure session without mic/speaker");
  console.log("  -h, --help                  Show this help text");
}

function parseArguments(argv) {
  const parsed = {
    apiKey: process.env.AZURE_VOICELIVE_API_KEY,
    endpoint: process.env.AZURE_VOICELIVE_ENDPOINT,
    model: process.env.AZURE_VOICELIVE_MODEL ?? "gpt-realtime",
    voice:
      process.env.AZURE_VOICELIVE_VOICE ?? "en-US-Ava:DragonHDLatestNeural",
    instructions:
      process.env.AZURE_VOICELIVE_INSTRUCTIONS ??
      "You are a helpful AI assistant. Respond naturally and conversationally. Keep your responses concise but engaging.",
    audioInputDevice: process.env.AUDIO_INPUT_DEVICE,
    listAudioDevices: false,
    greetingText: undefined,
    useTokenCredential: false,
    noAudio: false,
    help: false,
  };

  for (let i = 0; i < argv.length; i++) {
    const arg = argv[i];
    switch (arg) {
      case "--api-key":
        parsed.apiKey = argv[++i];
        break;
      case "--endpoint":
        parsed.endpoint = argv[++i];
        break;
      case "--model":
        parsed.model = argv[++i];
        break;
      case "--voice":
        parsed.voice = argv[++i];
        break;
      case "--instructions":
        parsed.instructions = argv[++i];
        break;
      case "--audio-input-device":
        parsed.audioInputDevice = argv[++i];
        break;
      case "--list-audio-devices":
        parsed.listAudioDevices = true;
        break;
      case "--greeting-text":
        parsed.greetingText = argv[++i];
        break;
      case "--use-token-credential":
        parsed.useTokenCredential = true;
        break;
      case "--no-audio":
        parsed.noAudio = true;
        break;
      case "--help":
      case "-h":
        parsed.help = true;
        break;
      default:
        if (arg?.startsWith("-")) {
          throw new Error(`Unknown option: ${arg}`);
        }
        break;
    }
  }

  return parsed;
}

/**
 * List available audio input devices on Windows (AudioEndpoint via WMI).
 * Falls back to a note on non-Windows platforms.
 */
async function listAudioDevices() {
  if (process.platform !== "win32") {
    console.log("Device listing is currently supported on Windows only.");
    console.log("On macOS/Linux, run: sox -V6 -n -t coreaudio -n trim 0 0  (or similar)");
    return;
  }

  const { execSync } = await import("node:child_process");
  try {
    const output = execSync(
      'powershell -NoProfile -Command "Get-CimInstance Win32_PnPEntity | Where-Object { $_.PNPClass -eq \'AudioEndpoint\' } | Select-Object -ExpandProperty Name"',
      { encoding: "utf-8", timeout: 10000 },
    ).trim();

    if (!output) {
      console.log("No audio endpoint devices found.");
      return;
    }

    console.log("Available audio endpoint devices:");
    console.log("");
    for (const line of output.split(/\r?\n/)) {
      const name = line.trim();
      if (name) console.log(`  ${name}`);
    }
    console.log("");
    console.log("Use the device name (or a unique substring) with --audio-input-device.");
    console.log('Example: node model-quickstart.js --audio-input-device "Microphone"');
  } catch (err) {
    console.error("Failed to query audio devices:", err.message);
  }
}

function resolveVoiceConfig(voiceName) {
  const looksLikeAzureVoice = voiceName.includes("-") || voiceName.includes(":");
  if (looksLikeAzureVoice) {
    return {
      type: "azure-standard",
      name: voiceName,
    };
  }

  return {
    type: "openai",
    name: voiceName,
  };
}

class AudioProcessor {
  constructor(enableAudio = true, inputDevice = undefined) {
    this._enableAudio = enableAudio;
    this._inputDevice = inputDevice;
    this._recorder = null;
    this._soxProcess = null;
    this._speaker = null;
    this._skipSeq = 0;
    this._nextSeq = 0;
    this._recordModule = null;
    this._speakerCtor = null;
  }

  async _ensureAudioModulesLoaded() {
    if (!this._enableAudio) return;
    if (this._recordModule && this._speakerCtor) return;

    try {
      const recordModule = await import("node-record-lpcm16");
      const speakerModule = await import("speaker");
      this._recordModule = recordModule.default;
      this._speakerCtor = speakerModule.default;
    } catch {
      throw new Error(
        "Audio dependencies are unavailable. Install optional packages (node-record-lpcm16, speaker) and required native build tools, or run with --no-audio for connectivity-only validation.",
      );
    }
  }

  async startCapture(session) {
    if (!this._enableAudio) {
      console.log("[audio] --no-audio enabled: microphone capture skipped");
      return;
    }
    if (this._recorder || this._soxProcess) return;

    if (this._inputDevice) {
      console.log(`[audio] Using explicit input device: ${this._inputDevice}`);
      const soxArgs = [
        "-q", "-t", "waveaudio", this._inputDevice,
        "-r", "24000", "-c", "1",
        "-e", "signed-integer", "-b", "16",
        "-t", "raw", "-",
      ];

      this._soxProcess = spawn("sox", soxArgs, {
        stdio: ["ignore", "pipe", "pipe"],
      });

      this._soxProcess.stdout.on("data", (chunk) => {
        if (session.isConnected) {
          session.sendAudio(new Uint8Array(chunk)).catch(() => {
            // Ignore send errors during disconnect
          });
        }
      });

      this._soxProcess.stderr.on("data", (data) => {
        const msg = data.toString().trim();
        if (msg) {
          console.error(`[audio] sox stderr: ${msg}`);
        }
      });

      this._soxProcess.on("error", (error) => {
        console.error(`[audio] SoX process error: ${error?.message ?? error}`);
      });

      this._soxProcess.on("close", (code) => {
        if (code !== 0) {
          console.error(`[audio] SoX exited with code ${code}`);
        }
        this._soxProcess = null;
      });

      console.log("[audio] Microphone capture started");
      return;
    }

    await this._ensureAudioModulesLoaded();

    const recorderOptions = {
      sampleRate: 24000,
      channels: 1,
      audioType: "raw",
      recorder: "sox",
      encoding: "signed-integer",
      bitwidth: 16,
    };

    this._recorder = this._recordModule.record(recorderOptions);
    const recorderStream = this._recorder.stream();

    recorderStream.on("data", (chunk) => {
      if (session.isConnected) {
        session.sendAudio(new Uint8Array(chunk)).catch(() => {
          // Ignore send errors during disconnect
        });
      }
    });

    recorderStream.on("error", (error) => {
      console.error(`[audio] Recorder stream error: ${error?.message ?? error}`);
      console.error(
        "[audio] SoX capture failed. Check microphone permissions/device and run with DEBUG=record for details.",
      );
    });

    console.log("[audio] Microphone capture started");
  }

  async startPlayback() {
    if (!this._enableAudio) {
      console.log("[audio] --no-audio enabled: speaker playback skipped");
      return;
    }
    if (this._speaker) return;
    await this._resetSpeaker();
    console.log("[audio] Playback ready");
  }

  queueAudio(base64Delta) {
    const seq = this._nextSeq++;
    if (seq < this._skipSeq) return;
    const chunk = Buffer.from(base64Delta, "base64");
    if (this._speaker && !this._speaker.destroyed) {
      this._speaker.write(chunk);
    }
  }

  skipPendingAudio() {
    if (!this._enableAudio) return;
    this._skipSeq = this._nextSeq++;
    this._resetSpeaker().catch(() => {
      // best-effort reset
    });
  }

  shutdown() {
    if (this._soxProcess) {
      try {
        this._soxProcess.kill();
      } catch {
        // no-op
      }
      this._soxProcess = null;
    }

    if (this._recorder) {
      this._recorder.stop();
      this._recorder = null;
    }

    if (this._speaker) {
      this._speaker.end();
      this._speaker = null;
    }

    console.log("[audio] Audio processor shut down");
  }

  async _resetSpeaker() {
    await this._ensureAudioModulesLoaded();

    if (this._speaker && !this._speaker.destroyed) {
      try {
        this._speaker.end();
      } catch {
        // no-op
      }
    }

    this._speaker = new this._speakerCtor({
      channels: 1,
      bitDepth: 16,
      sampleRate: 24000,
      signed: true,
    });

    this._speaker.on("error", () => {
      // Swallow transient audio device errors
    });
  }
}

class BasicModelVoiceAssistant {
  constructor(options) {
    this.endpoint = options.endpoint;
    this.credential = options.credential;
    this.model = options.model;
    this.voice = options.voice;
    this.instructions = options.instructions;
    this.audioInputDevice = options.audioInputDevice;
    this.greetingText = options.greetingText;
    this.noAudio = options.noAudio;

    this._session = null;
    this._subscription = null;
    this._audio = new AudioProcessor(!options.noAudio, options.audioInputDevice);
    this._activeResponse = false;
    this._responseApiDone = false;
    this._greetingSent = false;
  }

  async start() {
    const client = new VoiceLiveClient(this.endpoint, this.credential);
    const session = client.createSession({ model: this.model });
    this._session = session;

    console.log(
      `[init] Connecting to VoiceLive with model "${this.model}" at "${this.endpoint}" ...`,
    );

    this._subscription = session.subscribe({
      onSessionUpdated: async (event, context) => {
        const s = event.session;
        const model = s?.model;
        const voice = s?.voice;

        console.log(`[session] Session ready: ${context.sessionId}`);
        writeConversationLog(
          [
            `SessionID: ${context.sessionId}`,
            `Model: ${typeof model === "string" ? model : model?.toString?.() ?? ""}`,
            `Voice Name: ${voice?.name ?? ""}`,
            `Voice Type: ${voice?.type ?? ""}`,
            "",
          ].join("\n"),
        );

        if (!this._greetingSent) {
          this._greetingSent = true;
        }
      },

      onConversationItemInputAudioTranscriptionCompleted: async (event) => {
        const transcript = event.transcript ?? "";
        console.log(`👤 You said:\t${transcript}`);
        writeConversationLog(`User Input:\t${transcript}`);
      },

      onResponseTextDone: async (event) => {
        const text = event.text ?? "";
        console.log(`🤖 Assistant text:\t${text}`);
        writeConversationLog(`Assistant Text Response:\t${text}`);
      },

      onResponseAudioTranscriptDone: async (event) => {
        const transcript = event.transcript ?? "";
        console.log(`🤖 Assistant audio transcript:\t${transcript}`);
        writeConversationLog(`Assistant Audio Response:\t${transcript}`);
      },

      onInputAudioBufferSpeechStarted: async () => {
        console.log("🎤 Listening...");
        this._audio.skipPendingAudio();

        if (this._activeResponse && !this._responseApiDone) {
          try {
            await session.sendEvent({ type: "response.cancel" });
          } catch (err) {
            const msg = err?.message ?? "";
            if (!msg.toLowerCase().includes("no active response")) {
              console.warn("[barge-in] Cancel failed:", msg);
            }
          }
        }
      },

      onInputAudioBufferSpeechStopped: async () => {
        console.log("🤔 Processing...");
      },

      onResponseCreated: async () => {
        this._activeResponse = true;
        this._responseApiDone = false;
      },

      onResponseAudioDelta: async (event) => {
        if (event.delta) {
          this._audio.queueAudio(event.delta);
        }
      },

      onResponseAudioDone: async () => {
        console.log("🎤 Ready for next input...");
      },

      onResponseDone: async () => {
        console.log("✅ Response complete");
        this._activeResponse = false;
        this._responseApiDone = true;
      },

      onServerError: async (event) => {
        const msg = event.error?.message ?? "";
        if (msg.includes("Cancellation failed: no active response")) {
          return;
        }
        console.error(`❌ VoiceLive error: ${msg}`);
      },

      onConversationItemCreated: async (event) => {
        console.log(`[event] Conversation item created: ${event.item?.id ?? ""}`);
      },
    });

    await session.connect();
    console.log("[init] Connected to VoiceLive session websocket");

    await this._setupSession();

    if (!this._greetingSent) {
      this._greetingSent = true;
      await this._sendProactiveGreeting();
    }

    await this._audio.startPlayback();
    await this._audio.startCapture(session);

    console.log("\n" + "=".repeat(60));
    console.log("🎤 VOICE ASSISTANT READY");
    console.log("Start speaking to begin conversation");
    console.log("Press Ctrl+C to exit");
    console.log("=".repeat(60) + "\n");

    if (this.noAudio) {
      setTimeout(() => {
        process.emit("SIGINT");
      }, 6000);
    }

    await new Promise((resolve) => {
      const onSignal = () => resolve();
      process.once("SIGINT", onSignal);
      process.once("SIGTERM", onSignal);

      const poll = setInterval(() => {
        if (!session.isConnected) {
          clearInterval(poll);
          resolve();
        }
      }, 500);
    });

    await this.shutdown();
  }

  /**
   * Send a proactive greeting when the session starts.
   * Supports pre-defined (--greeting-text) or LLM-generated (default).
   */
  async _sendProactiveGreeting() {
    const session = this._session;

    if (this.greetingText) {
      console.log("[session] Sending pre-generated greeting ...");
      try {
        await session.sendEvent({
          type: "response.create",
          response: {
            preGeneratedAssistantMessage: {
              content: [{ type: "text", text: this.greetingText }],
            },
          },
        });
      } catch (err) {
        console.error("[session] Failed to send pre-generated greeting:", err.message);
      }
    } else {
      console.log("[session] Sending proactive greeting ...");
      try {
        await session.addConversationItem({
          type: "message",
          role: "system",
          content: [
            {
              type: "input_text",
              text: "Say something to welcome the user in English.",
            },
          ],
        });
        await session.sendEvent({ type: "response.create" });
      } catch (err) {
        console.error("[session] Failed to send greeting:", err.message);
      }
    }
  }

  async _setupSession() {
    console.log("[session] Configuring session ...");

    await this._session.updateSession({
      model: this.model,
      modalities: ["text", "audio"],
      instructions: this.instructions,
      voice: resolveVoiceConfig(this.voice),
      inputAudioFormat: "pcm16",
      outputAudioFormat: "pcm16",
      turnDetection: {
        type: "server_vad",
        threshold: 0.5,
        prefixPaddingInMs: 300,
        silenceDurationInMs: 500,
      },
      inputAudioEchoCancellation: { type: "server_echo_cancellation" },
      inputAudioNoiseReduction: { type: "azure_deep_noise_suppression" },
      inputAudioTranscription: { model: "azure-speech" },
    });

    console.log("[session] Session configuration sent");
  }

  async shutdown() {
    if (this._subscription) {
      await this._subscription.close();
      this._subscription = null;
    }

    if (this._session) {
      try {
        await this._session.disconnect();
      } catch {
        // ignore disconnect errors during shutdown
      }

      this._audio.shutdown();

      try {
        await this._session.dispose();
      } catch {
        // ignore dispose errors during shutdown
      }

      this._session = null;
    }
  }
}

async function main() {
  let args;
  try {
    args = parseArguments(process.argv.slice(2));
  } catch (err) {
    console.error(`❌ ${err.message}`);
    printUsage();
    process.exit(1);
  }

  if (args.help) {
    printUsage();
    return;
  }

  if (args.listAudioDevices) {
    await listAudioDevices();
    return;
  }

  if (!args.endpoint) {
    console.error(
      "❌ Missing endpoint. Set AZURE_VOICELIVE_ENDPOINT or pass --endpoint.",
    );
    process.exit(1);
  }

  if (!args.apiKey && !args.useTokenCredential) {
    console.error("❌ No authentication provided.");
    console.error(
      "Provide --api-key / AZURE_VOICELIVE_API_KEY or use --use-token-credential.",
    );
    process.exit(1);
  }

  const credential = args.useTokenCredential
    ? new DefaultAzureCredential()
    : new AzureKeyCredential(args.apiKey);

  console.log("Configuration:");
  console.log(`  AZURE_VOICELIVE_ENDPOINT: ${args.endpoint}`);
  console.log(`  AZURE_VOICELIVE_MODEL: ${args.model}`);
  console.log(`  AZURE_VOICELIVE_VOICE: ${args.voice}`);
  console.log(`  AUDIO_INPUT_DEVICE: ${args.audioInputDevice ?? "(not set)"}`);
  if (args.greetingText) {
    console.log(`  Proactive greeting: pre-defined`);
  } else {
    console.log(`  Proactive greeting: LLM-generated (default)`);
  }
  console.log(`  No audio mode: ${args.noAudio ? "enabled" : "disabled"}`);
  console.log(
    `  Authentication: ${args.useTokenCredential ? "DefaultAzureCredential" : "API Key"}`,
  );

  const assistant = new BasicModelVoiceAssistant({
    endpoint: args.endpoint,
    credential,
    model: args.model,
    voice: args.voice,
    instructions: args.instructions,
    audioInputDevice: args.audioInputDevice,
    greetingText: args.greetingText,
    noAudio: args.noAudio,
  });

  try {
    await assistant.start();
  } catch (err) {
    if (err?.code === "ERR_USE_AFTER_CLOSE") return;
    console.error("Fatal error:", err);
    process.exit(1);
  }
}

console.log("🎙️  Basic Voice Assistant with Azure VoiceLive SDK (Model Mode)");
console.log("=".repeat(60));
main().then(
  () => console.log("\n👋 Voice assistant shut down. Goodbye!"),
  (err) => {
    console.error("Unhandled error:", err);
    process.exit(1);
  },
);

  2. 运行语音助理：

    node model-quickstart.js

    若要使用Microsoft Entra ID身份验证而不是 API 密钥，请先登录并添加 --use-token-credential 标志：

    az login
node model-quickstart.js --use-token-credential

  3. 开始使用模型说话并听到响应。 可以通过说话来中断模型。 输入“Ctrl+C”退出对话。

输出

脚本的输出将打印到控制台。 你会看到指示连接状态、音频流和播放的消息。 音频通过扬声器或耳机播放。

🎙️  Basic Voice Assistant with Azure VoiceLive SDK (Model Mode)
============================================================

Configuration:
  AZURE_VOICELIVE_ENDPOINT: https://<your-resource>.cognitiveservices.azure.com/
  AZURE_VOICELIVE_MODEL: gpt-realtime
  AZURE_VOICELIVE_VOICE: en-US-Ava:DragonHDLatestNeural
  AUDIO_INPUT_DEVICE: (not set)
  Proactive greeting: LLM-generated (default)
  No audio mode: disabled
  Authentication: API Key

============================================================
🎤 VOICE ASSISTANT READY
Start speaking to begin conversation
Press Ctrl+C to exit
============================================================

🎤 Listening...
🤔 Processing...
👤 You said:	Hello.
🎤 Ready for next input...
🤖 Assistant text:	Hello! How can I assist you today?
🎤 Listening...
🤔 Processing...
👤 You said:	What's the capital/major city of France?
🎤 Ready for next input...
🤖 Assistant text:	The capital/major city of France is Paris. It's one of the most visited cities in the world, known for landmarks like the Eiffel Tower, the Louvre museum, and Notre-Dame Cathedral. Is there anything else you'd like to know?

👋 Voice assistant shut down. Goodbye!

在logs文件夹中创建了一个名为conversation_YYYYMMDD_HHmmss.log的对话日志文件。 此文件包含会话元数据和会话脚本。

SessionID: sess_2n3aSMKTRqkKqacFPzRqUM
Model: gpt-realtime
Voice Name: en-US-Ava:DragonHDLatestNeural
Voice Type: azure-standard

User Input:	Hello.
Assistant Text Response:	Hello! How can I assist you today?
User Input:	What's the capital/major city of France?
Assistant Text Response:	The capital of France is Paris...

以下是 技术日志 与 聊天日志之间的主要区别：

方面 对话日志 技术日志
观众 业务用户、内容审阅者 开发人员、IT运维
Content 谈话内容 系统工作原理
级别 应用程序/对话级别 系统/基础结构级别
故障排除 “模型说了什么？ “连接为何失败？”

示例：如果你的助手没有响应，你将检查：

  • 控制台日志 →“WebSocket 连接失败”或“音频流错误”
  • 对话日志 →“用户是否真的说了什么？”

这两个日志都是互补的 - 用于聊天分析和测试的对话日志、系统诊断的技术日志。

技术日志

目的：技术调试和系统监视

内容：

  • WebSocket 连接事件
  • 音频流状态
  • 错误消息和堆栈跟踪
  • 系统级事件（session.created、response.done 等）
  • 网络连接问题
  • 音频处理诊断

格式：带有括号前缀的控制台输出（例如，、 [session][audio][init]）

用例：

  • 调试连接问题
  • 监视系统性能
  • 排查音频故障
  • 开发/运维分析

对话日志

目的：对话脚本和用户体验跟踪

内容：

  • 模型和会话标识
  • 会话配置详细信息
  • 用户脚本：“今天天气如何？”，“停止”
  • 模型响应：完整响应文本
  • 聊天流和交互

格式：纯文本、人工可读对话格式

用例：

  • 分析对话质量
  • 回顾实际所说的内容
  • 了解用户交互和模型响应
  • 业务/内容分析

本文介绍如何使用 voiceLive SDK for Java 将 Voice Live 与 Microsoft Foundry 模型配合使用。

Reference 文档 | 包（Maven） | GitHub 上的更多示例

创建并运行一个应用程序，将 Voice Live 直接与生成型 AI 模型结合使用，以创建实时语音代理。

  • 使用模型直接允许为每个会话指定自定义指令（提示），为动态或实验用例提供更大的灵活性。

  • 如果希望对会话参数进行精细控制，或者需要频繁调整提示或配置，而无需在门户中更新代理，则模型可能更可取。

  • 基于模型的会话的代码在某些方面更简单，因为它不需要管理代理 ID 或特定于代理的设置。

  • 直接模型使用适用于不需要代理级抽象或内置逻辑的方案。

若要改用语音实时 API 和代理，请参阅 语音实时 API 代理快速入门。

先决条件

小窍门

若要使用语音直播，无需使用 Foundry 资源部署音频模型。 Voice Live 是全托管的，模型会自动为你部署。 有关模型可用性的详细信息，请参阅 Voice Live 概述文档。

注释

若要使用 Microsoft Entra ID 进行无密钥身份验证，请安装 Azure CLI并将 Cognitive Services User 角色分配给用户帐户。 可以在 Azure 门户中的 访问控制（IAM）>添加角色分配下分配角色。

设置

  1. 创建新文件夹 voice-live-quickstart，并使用以下命令转到快速入门文件夹：

    mkdir voice-live-quickstart && cd voice-live-quickstart

  2. 在项目目录的根目录中，创建一个pom.xml文件，并使用以下内容：

    <?xml version="1.0" encoding="UTF-8"?>
<project xmlns="http://maven.apache.org/POM/4.0.0"
         xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
         xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 http://maven.apache.org/xsd/maven-4.0.0.xsd">
    <modelVersion>4.0.0</modelVersion>

    <groupId>com.azure.ai.voicelive</groupId>
    <artifactId>model-quickstart</artifactId>
    <version>1.0.0</version>
    <packaging>jar</packaging>

    <name>Azure VoiceLive Model Quickstart</name>
    <description>Model quickstart sample for Azure AI VoiceLive SDK</description>

    <properties>
        <maven.compiler.source>11</maven.compiler.source>
        <maven.compiler.target>11</maven.compiler.target>
        <project.build.sourceEncoding>UTF-8</project.build.sourceEncoding>
    </properties>

    <dependencies>
        <!-- Azure VoiceLive SDK -->
        <dependency>
            <groupId>com.azure</groupId>
            <artifactId>azure-ai-voicelive</artifactId>
            <version>1.0.0-beta.1</version>
        </dependency>

        <!-- Azure Core -->
        <dependency>
            <groupId>com.azure</groupId>
            <artifactId>azure-core</artifactId>
            <version>1.53.0</version>
        </dependency>

        <!-- Azure Identity for authentication -->
        <dependency>
            <groupId>com.azure</groupId>
            <artifactId>azure-identity</artifactId>
            <version>1.11.0</version>
        </dependency>

        <!-- Reactor Core for reactive programming -->
        <dependency>
            <groupId>io.projectreactor</groupId>
            <artifactId>reactor-core</artifactId>
            <version>3.5.11</version>
        </dependency>

        <!-- SLF4J for logging -->
        <dependency>
            <groupId>org.slf4j</groupId>
            <artifactId>slf4j-api</artifactId>
            <version>2.0.9</version>
        </dependency>
        <dependency>
            <groupId>org.slf4j</groupId>
            <artifactId>slf4j-simple</artifactId>
            <version>2.0.9</version>
        </dependency>
    </dependencies>

    <build>
        <sourceDirectory>.</sourceDirectory>
        <plugins>
            <plugin>
                <groupId>org.apache.maven.plugins</groupId>
                <artifactId>maven-compiler-plugin</artifactId>
                <version>3.11.0</version>
                <configuration>
                    <source>11</source>
                    <target>11</target>
                </configuration>
            </plugin>
            <plugin>
                <groupId>org.codehaus.mojo</groupId>
                <artifactId>exec-maven-plugin</artifactId>
                <version>3.1.0</version>
                <configuration>
                    <mainClass>ModelQuickstart</mainClass>
                </configuration>
            </plugin>
        </plugins>
    </build>
</project>

    注释

    <sourceDirectory>.</sourceDirectory> 配置指示 Maven 在当前目录中查找Java源文件，而不是默认的 src/main/java 结构。 这允许更简单的平面项目结构。

  3. 安装依赖项：

    mvn clean install

  4. 配置身份验证 - 将application.properties.sample中的内容复制到application.properties并根据你的数值进行更新：

    azure.voicelive.endpoint=https://your-resource-name.services.ai.azure.com/
azure.voicelive.api-key=your-api-key
azure.voicelive.api-version=2025-10-01

    注释

    还可以使用环境变量，而不是 application.properties。 设置 AZURE_VOICELIVE_ENDPOINT 和 AZURE_VOICELIVE_API_KEY。 应用程序将首先检查 application.properties ，然后回退到环境变量。

  5. 运行示例：

    mvn exec:java

    若要使用Azure令牌凭据身份验证，而不是 API 密钥：

    az login
mvn exec:java -Dexec.args="--use-token-credential"

    注释

    在一些终端（例如 PowerShell）中，你可能需要对参数进行转义。 在 PowerShell 中使用 mvn exec:java `"-Dexec.args=--use-token-credential`"

检索资源信息

在要在其中运行代码的文件夹中创建一 .env 个名为的新文件。

在 .env 文件中，添加以下用于身份验证的环境变量：

AZURE_VOICELIVE_ENDPOINT=<your_endpoint>
AZURE_VOICELIVE_MODEL=<your_model>
AZURE_VOICELIVE_API_VERSION=2025-10-01
AZURE_VOICELIVE_API_KEY=<your_api_key> # Only required if using API key authentication

将默认值替换为实际终结点、模型、API 版本和 API 密钥。

变量名称 价值
AZURE_VOICELIVE_ENDPOINT 从 Azure 门户检查资源时，可以在 Keys 和 Endpoint 节中找到此值。
AZURE_VOICELIVE_MODEL 要使用的模型。 例如，gpt-4o 或 gpt-realtime-mini。 有关模型可用性的详细信息，请参阅 Voice Live API 概述文档。
AZURE_VOICELIVE_API_VERSION 要使用的 API 版本。 例如，2025-10-01。

详细了解无密钥身份验证，以及如何设置环境变量。

添加示例代码

使用以下代码创建 ModelQuickstart.java 文件：

// Copyright (c) Microsoft Corporation. All rights reserved.
// Licensed under the MIT License.

import com.azure.ai.voicelive.VoiceLiveAsyncClient;
import com.azure.ai.voicelive.VoiceLiveClientBuilder;
import com.azure.ai.voicelive.VoiceLiveServiceVersion;
import com.azure.ai.voicelive.VoiceLiveSessionAsyncClient;
import com.azure.ai.voicelive.models.AudioEchoCancellation;
import com.azure.ai.voicelive.models.AudioInputTranscriptionOptions;
import com.azure.ai.voicelive.models.AudioInputTranscriptionOptionsModel;
import com.azure.ai.voicelive.models.AudioNoiseReduction;
import com.azure.ai.voicelive.models.AudioNoiseReductionType;
import com.azure.ai.voicelive.models.ClientEventSessionUpdate;
import com.azure.ai.voicelive.models.InputAudioFormat;
import com.azure.ai.voicelive.models.InteractionModality;
import com.azure.ai.voicelive.models.AzureStandardVoice;
import com.azure.ai.voicelive.models.OutputAudioFormat;
import com.azure.ai.voicelive.models.ServerEventType;
import com.azure.ai.voicelive.models.ServerVadTurnDetection;
import com.azure.ai.voicelive.models.SessionUpdate;
import com.azure.ai.voicelive.models.SessionUpdateError;
import com.azure.ai.voicelive.models.SessionUpdateResponseAudioDelta;
import com.azure.ai.voicelive.models.SessionUpdateSessionUpdated;
import com.azure.ai.voicelive.models.VoiceLiveSessionOptions;
import com.azure.core.credential.KeyCredential;
import com.azure.core.credential.TokenCredential;
import com.azure.core.util.BinaryData;
import com.azure.identity.AzureCliCredentialBuilder;
import reactor.core.publisher.Mono;
import reactor.core.scheduler.Schedulers;

import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.TargetDataLine;

import java.io.FileInputStream;
import java.io.IOException;
import java.io.InputStream;
import java.util.Arrays;
import java.util.Properties;
import java.util.concurrent.BlockingQueue;
import java.util.concurrent.LinkedBlockingQueue;
import java.util.concurrent.atomic.AtomicBoolean;
import java.util.concurrent.atomic.AtomicInteger;
import java.util.concurrent.atomic.AtomicReference;

/**
    * Complete voice assistant sample demonstrating full-featured real-time voice conversation.
    *
    * <p><strong>NOTE:</strong> This is a comprehensive sample showing all features together.
    * For easier understanding, see these focused samples:</p>
    * <ul>
    *   <li>{@link BasicVoiceConversationSample} - Minimal setup and session management</li>
    *   <li>{@link MicrophoneInputSample} - Audio capture from microphone</li>
    *   <li>{@link AudioPlaybackSample} - Audio playback to speakers</li>
    *   <li>{@link AuthenticationMethodsSample} - Different authentication methods</li>
    * </ul>
    *
    * <p>This sample demonstrates:</p>
    * <ul>
    *   <li>Real-time microphone audio capture</li>
    *   <li>Streaming audio to VoiceLive service</li>
    *   <li>Receiving and playing audio responses</li>
    *   <li>Voice Activity Detection (VAD) with interruption handling</li>
    *   <li>Multi-threaded audio processing</li>
    *   <li>Audio transcription with Whisper</li>
    *   <li>Noise reduction and echo cancellation</li>
    *   <li>Dual authentication support (API key and token credential)</li>
    * </ul>
    *
    * <p><strong>Environment Variables Required:</strong></p>
    * <ul>
    *   <li>AZURE_VOICELIVE_ENDPOINT - The VoiceLive service endpoint URL</li>
    *   <li>AZURE_VOICELIVE_API_KEY - The API key (required if not using --use-token-credential)</li>
    * </ul>
    *
    * <p><strong>Audio Requirements:</strong></p>
    * The sample requires a working microphone and speakers/headphones.
    * Audio format is 24kHz, 16-bit PCM, mono as required by the VoiceLive service.
    *
    * <p><strong>How to Run:</strong></p>
    * <pre>{@code
    * # With API Key (default):
    * mvn exec:java -Dexec.mainClass="com.azure.ai.voicelive.VoiceAssistantSample" -Dexec.classpathScope=test
    *
    * # With Token Credential:
    * mvn exec:java -Dexec.mainClass="ModelQuickstart" -Dexec.classpathScope=test -Dexec.args="--use-token-credential"
    * }</pre>
    */
public final class ModelQuickstart {

    // Service configuration constants
    private static final String DEFAULT_API_VERSION = "2025-10-01";
    private static final String DEFAULT_MODEL = "gpt-realtime";
    private static final String DEFAULT_VOICE = "en-US-Ava:DragonHDLatestNeural";
    private static final String DEFAULT_INSTRUCTIONS = "You are a helpful AI voice assistant. Respond naturally and conversationally. Keep your responses concise but engaging. Speak as if having a real conversation.";

    // Environment variable names
    private static final String ENV_ENDPOINT = "AZURE_VOICELIVE_ENDPOINT";
    private static final String ENV_API_KEY = "AZURE_VOICELIVE_API_KEY";

    // Audio format constants (VoiceLive requirements)
    private static final int SAMPLE_RATE = 24000;          // 24kHz as required by VoiceLive
    private static final int CHANNELS = 1;                 // Mono
    private static final int SAMPLE_SIZE_BITS = 16;        // 16-bit PCM
    private static final int CHUNK_SIZE = 1200;            // 50ms chunks (24000 * 0.05)
    private static final int AUDIO_BUFFER_SIZE_MULTIPLIER = 4;

    // Private constructor to prevent instantiation
    private ModelQuickstart() {
        throw new UnsupportedOperationException("Utility class cannot be instantiated");
    }

    /**
        * Audio packet for playback queue management.
        * Uses sequence numbers to support interruption handling.
        */
    private static class AudioPlaybackPacket {
        final int sequenceNumber;
        final byte[] audioData;

        AudioPlaybackPacket(int sequenceNumber, byte[] audioData) {
            this.sequenceNumber = sequenceNumber;
            this.audioData = audioData;
        }
    }

    /**
        * Handles real-time audio capture from microphone and playback to speakers.
        *
        * <p>This class manages two separate threads:</p>
        * <ul>
        *   <li>Capture thread: Continuously reads audio from microphone and sends to VoiceLive service</li>
        *   <li>Playback thread: Receives audio responses and plays them through speakers</li>
        * </ul>
        *
        * <p>Supports interruption handling where user speech can cancel ongoing assistant responses.</p>
        */
    private static class AudioProcessor {
        private final VoiceLiveSessionAsyncClient session;
        private final AudioFormat audioFormat;

        // Audio capture components
        private TargetDataLine microphone;
        private final AtomicBoolean isCapturing = new AtomicBoolean(false);

        // Audio playback components
        private SourceDataLine speaker;
        private final BlockingQueue<AudioPlaybackPacket> playbackQueue = new LinkedBlockingQueue<>();
        private final AtomicBoolean isPlaying = new AtomicBoolean(false);
        private final AtomicInteger nextSequenceNumber = new AtomicInteger(0);
        private final AtomicInteger playbackBase = new AtomicInteger(0);

        AudioProcessor(VoiceLiveSessionAsyncClient session) {
            this.session = session;
            this.audioFormat = new AudioFormat(
                AudioFormat.Encoding.PCM_SIGNED,
                SAMPLE_RATE,
                SAMPLE_SIZE_BITS,
                CHANNELS,
                CHANNELS * SAMPLE_SIZE_BITS / 8, // frameSize
                SAMPLE_RATE,
                false // bigEndian
            );
        }

        /**
            * Start capturing audio from microphone
            */
        void startCapture() {
            if (isCapturing.get()) {
                return;
            }

            try {
                DataLine.Info micInfo = new DataLine.Info(TargetDataLine.class, audioFormat);

                if (!AudioSystem.isLineSupported(micInfo)) {
                    throw new UnsupportedOperationException("Microphone not supported with required format");
                }

                microphone = (TargetDataLine) AudioSystem.getLine(micInfo);
                microphone.open(audioFormat, CHUNK_SIZE * AUDIO_BUFFER_SIZE_MULTIPLIER);
                microphone.start();

                isCapturing.set(true);

                // Start capture thread
                Thread captureThread = new Thread(this::captureAudioLoop, "VoiceLive-AudioCapture");
                captureThread.setDaemon(true);
                captureThread.start();

                System.out.println("🎤 Microphone capture started");

            } catch (LineUnavailableException e) {
                System.err.println("❌ Failed to start microphone: " + e.getMessage());
                throw new RuntimeException("Failed to initialize microphone", e);
            }
        }

        /**
            * Starts audio playback system.
            */
        void startPlayback() {
            if (isPlaying.get()) {
                return;
            }

            try {
                DataLine.Info speakerInfo = new DataLine.Info(SourceDataLine.class, audioFormat);

                if (!AudioSystem.isLineSupported(speakerInfo)) {
                    throw new UnsupportedOperationException("Speaker not supported with required format");
                }

                speaker = (SourceDataLine) AudioSystem.getLine(speakerInfo);
                speaker.open(audioFormat, CHUNK_SIZE * AUDIO_BUFFER_SIZE_MULTIPLIER);
                speaker.start();

                isPlaying.set(true);

                // Start playback thread
                Thread playbackThread = new Thread(this::playbackAudioLoop, "VoiceLive-AudioPlayback");
                playbackThread.setDaemon(true);
                playbackThread.start();

                System.out.println("🔊 Audio playback started");

            } catch (LineUnavailableException e) {
                System.err.println("❌ Failed to start speaker: " + e.getMessage());
                throw new RuntimeException("Failed to initialize speaker", e);
            }
        }

        /**
            * Audio capture loop - runs in separate thread
            */
        private void captureAudioLoop() {
            byte[] buffer = new byte[CHUNK_SIZE * 2]; // 16-bit samples
            System.out.println("🎤 Audio capture loop started");

            while (isCapturing.get() && microphone != null) {
                try {
                    int bytesRead = microphone.read(buffer, 0, buffer.length);
                    if (bytesRead > 0) {
                        // Send audio to VoiceLive service
                        byte[] audioChunk = Arrays.copyOf(buffer, bytesRead);

                        // Send audio asynchronously using the session's audio buffer append
                        session.sendInputAudio(BinaryData.fromBytes(audioChunk))
                            .subscribeOn(Schedulers.boundedElastic())
                            .subscribe(
                                v -> {}, // onNext
                                error -> {
                                    // Only log non-interruption errors
                                    if (!error.getMessage().contains("cancelled")) {
                                        System.err.println("❌ Error sending audio: " + error.getMessage());
                                    }
                                }
                            );
                    }
                } catch (Exception e) {
                    if (isCapturing.get()) {
                        System.err.println("❌ Error in audio capture: " + e.getMessage());
                    }
                    break;
                }
            }
            System.out.println("🎤 Audio capture loop ended");
        }

        /**
            * Audio playback loop - runs in separate thread
            */
        private void playbackAudioLoop() {
            while (isPlaying.get()) {
                try {
                    AudioPlaybackPacket packet = playbackQueue.take(); // Blocking wait

                    if (packet.audioData == null) {
                        // Shutdown signal
                        break;
                    }

                    // Check if packet should be skipped (interrupted)
                    int currentBase = playbackBase.get();
                    if (packet.sequenceNumber < currentBase) {
                        // Skip interrupted audio
                        continue;
                    }

                    // Play the audio
                    if (speaker != null && speaker.isOpen()) {
                        speaker.write(packet.audioData, 0, packet.audioData.length);
                    }

                } catch (InterruptedException e) {
                    Thread.currentThread().interrupt();
                    break;
                } catch (Exception e) {
                    System.err.println("❌ Error in audio playback: " + e.getMessage());
                }
            }
        }

        /**
            * Queue audio data for playback
            */
        void queueAudio(byte[] audioData) {
            if (audioData != null && audioData.length > 0) {
                int seqNum = nextSequenceNumber.getAndIncrement();
                playbackQueue.offer(new AudioPlaybackPacket(seqNum, audioData));
            }
        }

        /**
            * Skip pending audio (for interruption handling)
            */
        void skipPendingAudio() {
            playbackBase.set(nextSequenceNumber.get());
            playbackQueue.clear();

            // Also drain the speaker buffer to stop playback immediately
            if (speaker != null && speaker.isOpen()) {
                speaker.flush();
            }
        }

        /**
            * Stop capture and playback
            */
        void shutdown() {
            // Stop capture
            isCapturing.set(false);
            if (microphone != null) {
                microphone.stop();
                microphone.close();
                microphone = null;
            }
            System.out.println("🎤 Microphone capture stopped");

            // Stop playback
            isPlaying.set(false);
            playbackQueue.offer(new AudioPlaybackPacket(-1, null)); // Shutdown signal
            if (speaker != null) {
                speaker.stop();
                speaker.close();
                speaker = null;
            }
            System.out.println("🔊 Audio playback stopped");
        }
    }

    /**
        * Configuration class to hold application settings.
        */
    private static class Config {
        String endpoint;
        String apiKey;
        String model = DEFAULT_MODEL;
        String voice = DEFAULT_VOICE;
        String instructions = DEFAULT_INSTRUCTIONS;
        boolean useTokenCredential = false;

        static Config load(String[] args) {
            Config config = new Config();
            
            // 1. Load from application.properties first
            Properties props = loadProperties();
            if (props != null) {
                config.endpoint = props.getProperty("azure.voicelive.endpoint");
                config.apiKey = props.getProperty("azure.voicelive.api-key");
                config.model = props.getProperty("azure.voicelive.model", DEFAULT_MODEL);
                config.voice = props.getProperty("azure.voicelive.voice", DEFAULT_VOICE);
                config.instructions = props.getProperty("azure.voicelive.instructions", DEFAULT_INSTRUCTIONS);
            }
            
            // 2. Override with environment variables if present
            if (System.getenv(ENV_ENDPOINT) != null) {
                config.endpoint = System.getenv(ENV_ENDPOINT);
            }
            if (System.getenv(ENV_API_KEY) != null) {
                config.apiKey = System.getenv(ENV_API_KEY);
            }
            if (System.getenv("AZURE_VOICELIVE_MODEL") != null) {
                config.model = System.getenv("AZURE_VOICELIVE_MODEL");
            }
            if (System.getenv("AZURE_VOICELIVE_VOICE") != null) {
                config.voice = System.getenv("AZURE_VOICELIVE_VOICE");
            }
            if (System.getenv("AZURE_VOICELIVE_INSTRUCTIONS") != null) {
                config.instructions = System.getenv("AZURE_VOICELIVE_INSTRUCTIONS");
            }
            
            // 3. Parse command line arguments (highest priority)
            for (int i = 0; i < args.length; i++) {
                switch (args[i]) {
                    case "--endpoint":
                        if (i + 1 < args.length) config.endpoint = args[++i];
                        break;
                    case "--api-key":
                        if (i + 1 < args.length) config.apiKey = args[++i];
                        break;
                    case "--model":
                        if (i + 1 < args.length) config.model = args[++i];
                        break;
                    case "--voice":
                        if (i + 1 < args.length) config.voice = args[++i];
                        break;
                    case "--instructions":
                        if (i + 1 < args.length) config.instructions = args[++i];
                        break;
                    case "--use-token-credential":
                        config.useTokenCredential = true;
                        break;
                }
            }
            
            return config;
        }
    }

    /**
        * Load configuration from application.properties file.
        */
    private static Properties loadProperties() {
        Properties props = new Properties();
        try (InputStream input = new FileInputStream("application.properties")) {
            props.load(input);
            System.out.println("✓ Loaded configuration from application.properties");
            return props;
        } catch (IOException e) {
            // File not found or cannot be read - this is OK, will use env vars
            return null;
        }
    }

    /**
        * Main method to run the voice assistant sample.
        *
        * <p>Configuration priority (highest to lowest):</p>
        * <ol>
        *   <li>Command line arguments</li>
        *   <li>Environment variables</li>
        *   <li>application.properties file</li>
        * </ol>
        *
        * <p>Supported command line arguments:</p>
        * <ul>
        *   <li>--endpoint &lt;url&gt; - VoiceLive endpoint URL</li>
        *   <li>--api-key &lt;key&gt; - API key for authentication</li>
        *   <li>--model &lt;model&gt; - Model to use (default: gpt-realtime)</li>
        *   <li>--voice &lt;voice&gt; - Voice name (e.g., en-US-Ava:DragonHDLatestNeural)</li>
        *   <li>--instructions &lt;text&gt; - Custom system instructions</li>
        *   <li>--use-token-credential - Use Azure CLI authentication instead of API key</li>
        * </ul>
        *
        * @param args Command line arguments
        */
    public static void main(String[] args) {
        // Load configuration
        Config config = Config.load(args);

        // Validate configuration
        if (config.endpoint == null) {
            printUsage();
            return;
        }

        if (!config.useTokenCredential && config.apiKey == null) {
            System.err.println("❌ API key is required when not using --use-token-credential");
            System.err.println("   Set it via:");
            System.err.println("   - application.properties: azure.voicelive.api-key=<your-key>");
            System.err.println("   - Environment variable: AZURE_VOICELIVE_API_KEY=<your-key>");
            System.err.println("   - Command line: --api-key <your-key>");
            printUsage();
            return;
        }

        // Check audio system availability
        if (!checkAudioSystem()) {
            System.err.println("❌ Audio system check failed. Please ensure microphone and speakers are available.");
            return;
        }

        System.out.println("🎙️ Starting Voice Assistant...");
        System.out.println("   Model: " + config.model);
        if (config.voice != null) {
            System.out.println("   Voice: " + config.voice);
        }

        try {
            if (config.useTokenCredential) {
                // Use token credential authentication (Azure CLI)
                System.out.println("🔑 Using Token Credential authentication (Azure CLI)");
                System.out.println("   Make sure you have run 'az login' before running this sample");
                TokenCredential credential = new AzureCliCredentialBuilder().build();
                runVoiceAssistant(config, credential);
            } else {
                // Use API Key authentication
                System.out.println("🔑 Using API Key authentication");
                runVoiceAssistant(config, new KeyCredential(config.apiKey));
            }
            System.out.println("✓ Voice Assistant completed successfully");
        } catch (Exception e) {
            System.err.println("❌ Voice Assistant failed: " + e.getMessage());
            e.printStackTrace();
        }
    }

    /**
        * Check if audio system is available
        */
    private static boolean checkAudioSystem() {
        try {
            AudioFormat format = new AudioFormat(SAMPLE_RATE, SAMPLE_SIZE_BITS, CHANNELS, true, false);

            // Check microphone
            DataLine.Info micInfo = new DataLine.Info(TargetDataLine.class, format);
            if (!AudioSystem.isLineSupported(micInfo)) {
                System.err.println("❌ No compatible microphone found");
                return false;
            }

            // Check speaker
            DataLine.Info speakerInfo = new DataLine.Info(SourceDataLine.class, format);
            if (!AudioSystem.isLineSupported(speakerInfo)) {
                System.err.println("❌ No compatible speaker found");
                return false;
            }

            System.out.println("✓ Audio system check passed");
            return true;

        } catch (Exception e) {
            System.err.println("❌ Audio system check failed: " + e.getMessage());
            return false;
        }
    }

    /**
        * Prints usage instructions for setting up environment variables.
        */
    private static void printUsage() {
        System.err.println("\n═══════════════════════════════════════════════════════════════");
        System.err.println("Usage: mvn exec:java [options]");
        System.err.println("═══════════════════════════════════════════════════════════════");
        System.err.println("\nConfiguration (in priority order):");
        System.err.println("  1. Command line arguments (--endpoint, --api-key, etc.)");
        System.err.println("  2. Environment variables (AZURE_VOICELIVE_ENDPOINT, etc.)");
        System.err.println("  3. application.properties file");
        System.err.println("\nCommand Line Options:");
        System.err.println("  --endpoint <url>         VoiceLive endpoint URL");
        System.err.println("  --api-key <key>          API key for authentication");
        System.err.println("  --model <model>          Model to use (default: gpt-realtime)");
        System.err.println("  --voice <voice>          Voice name (e.g., en-US-Ava:DragonHDLatestNeural)");
        System.err.println("  --instructions <text>    Custom system instructions");
        System.err.println("  --use-token-credential   Use Azure CLI authentication");
        System.err.println("\nExamples:");
        System.err.println("  # Using application.properties:");
        System.err.println("  mvn exec:java");
        System.err.println("\n  # Using command line arguments:");
        System.err.println("  mvn exec:java -Dexec.args=\"--endpoint https://... --api-key <key>\"");
        System.err.println("\n  # Using Azure CLI authentication:");
        System.err.println("  mvn exec:java -Dexec.args=\"--use-token-credential\"");
        System.err.println("\n  # With custom model and voice:");
        System.err.println("  mvn exec:java -Dexec.args=\"--model gpt-4.1 --voice en-US-JennyNeural\"");
        System.err.println("═══════════════════════════════════════════════════════════════\n");
    }

    /**
        * Run the voice assistant with API key authentication.
        *
        * @param config The configuration object
        * @param credential The API key credential
        */
    private static void runVoiceAssistant(Config config, KeyCredential credential) {
        System.out.println("🔧 Initializing VoiceLive client:");
        System.out.println("   Endpoint: " + config.endpoint);

        // Create the VoiceLive client
        VoiceLiveAsyncClient client = new VoiceLiveClientBuilder()
            .endpoint(config.endpoint)
            .credential(credential)
            .serviceVersion(VoiceLiveServiceVersion.V2025_10_01)
            .buildAsyncClient();

        runVoiceAssistantWithClient(client, config);
    }

    /**
        * Run the voice assistant with Azure AD authentication.
        *
        * @param config The configuration object
        * @param credential The token credential
        */
    private static void runVoiceAssistant(Config config, TokenCredential credential) {
        System.out.println("🔧 Initializing VoiceLive client:");
        System.out.println("   Endpoint: " + config.endpoint);

        // Create the VoiceLive client
        VoiceLiveAsyncClient client = new VoiceLiveClientBuilder()
            .endpoint(config.endpoint)
            .credential(credential)
            .serviceVersion(VoiceLiveServiceVersion.V2025_10_01)
            .buildAsyncClient();

        runVoiceAssistantWithClient(client, config);
    }

    /**
        * Run the voice assistant with the configured client.
        *
        * @param client The VoiceLive async client
        * @param config The configuration object
        */
    private static void runVoiceAssistantWithClient(VoiceLiveAsyncClient client, Config config) {
        System.out.println("✓ VoiceLive client created");

        // Configure session options for voice conversation
        VoiceLiveSessionOptions sessionOptions = createVoiceSessionOptions(config);
        AtomicReference<AudioProcessor> audioProcessorRef = new AtomicReference<>();

        // Execute the reactive workflow - start with the configured model
        client.startSession(config.model)
            .flatMap(session -> {
                System.out.println("✓ Session started successfully");

                // Create audio processor
                AudioProcessor audioProcessor = new AudioProcessor(session);
                audioProcessorRef.set(audioProcessor);

                // Subscribe to receive server events asynchronously
                session.receiveEvents()
                    .doOnSubscribe(subscription -> System.out.println("🔗 Subscribed to event stream"))
                    .doOnComplete(() -> System.out.println("⚠️ Event stream completed (this might indicate a connection issue)"))
                    .doOnError(error -> System.out.println("❌ Event stream error: " + error.getMessage()))
                    .subscribe(
                        event -> handleServerEvent(event, audioProcessor),
                        error -> System.err.println("❌ Error receiving events: " + error.getMessage()),
                        () -> System.out.println("✓ Event stream completed")
                    );

                System.out.println("📤 Sending session.update configuration...");
                ClientEventSessionUpdate updateEvent = new ClientEventSessionUpdate(sessionOptions);
                session.sendEvent(updateEvent)
                    .doOnSuccess(v -> System.out.println("✓ Session configuration sent"))
                    .doOnError(error -> System.err.println("❌ Failed to send session.update: " + error.getMessage()))
                    .subscribe();


                // Start audio systems
                audioProcessor.startPlayback();

                System.out.println("🎤 VOICE ASSISTANT READY");
                System.out.println("Start speaking to begin conversation");
                System.out.println("Press Ctrl+C to exit");

                // Install shutdown hook for graceful cleanup
                Runtime.getRuntime().addShutdownHook(new Thread(() -> {
                    System.out.println("\n🛑 Shutting down gracefully...");
                    audioProcessor.shutdown();
                }));

                // Keep the reactive chain alive to continue processing events
                // Mono.never() prevents the chain from completing, allowing the event stream to run
                // The shutdown hook above handles cleanup when the JVM exits (Ctrl+C)
                // Note: In production, use a proper signal mechanism (e.g., CountDownLatch, CompletableFuture)
                return Mono.never();
            })
            .doOnError(error -> System.err.println("❌ Error: " + error.getMessage()))
            .doFinally(signalType -> {
                // Cleanup audio processor
                AudioProcessor audioProcessor = audioProcessorRef.get();
                if (audioProcessor != null) {
                    audioProcessor.shutdown();
                }
            })
            .block(); // Block only for demo purposes; use reactive patterns in production
    }

    /**
        * Create session configuration for voice conversation
        */
    private static VoiceLiveSessionOptions createVoiceSessionOptions(Config config) {
        System.out.println("🔧 Creating session configuration:");

        // Create server VAD configuration similar to Python sample
        ServerVadTurnDetection turnDetection = new ServerVadTurnDetection()
            .setThreshold(0.5)
            .setPrefixPaddingMs(300)
            .setSilenceDurationMs(500)
            .setInterruptResponse(true)
            .setAutoTruncate(true)
            .setCreateResponse(true);

        // Create audio input transcription configuration
        AudioInputTranscriptionOptions transcriptionOptions = new AudioInputTranscriptionOptions(AudioInputTranscriptionOptionsModel.WHISPER_1);

        VoiceLiveSessionOptions options = new VoiceLiveSessionOptions()
            .setInstructions(config.instructions)
            // Voice: AzureStandardVoice for Azure TTS voices (e.g., en-US-Ava:DragonHDLatestNeural)
            .setVoice(BinaryData.fromObject(new AzureStandardVoice(config.voice)))
            .setModalities(Arrays.asList(InteractionModality.TEXT, InteractionModality.AUDIO))
            .setInputAudioFormat(InputAudioFormat.PCM16)
            .setOutputAudioFormat(OutputAudioFormat.PCM16)
            .setInputAudioSamplingRate(SAMPLE_RATE)
            .setInputAudioNoiseReduction(new AudioNoiseReduction(AudioNoiseReductionType.NEAR_FIELD))
            .setInputAudioEchoCancellation(new AudioEchoCancellation())
            .setInputAudioTranscription(transcriptionOptions)
            .setTurnDetection(turnDetection);


        System.out.println("✓ Session configuration created");
        return options;
    }

    /**
        * Handle incoming server events
        */
    private static void handleServerEvent(SessionUpdate event, AudioProcessor audioProcessor) {
        ServerEventType eventType = event.getType();

        try {
            if (eventType == ServerEventType.SESSION_CREATED) {
                System.out.println("✓ Session created - initializing...");
            } else if (eventType == ServerEventType.SESSION_UPDATED) {
                System.out.println("✓ Session updated - starting microphone");

                // Now that bufferObject() bug is fixed in generated code, we can access the typed class
                if (event instanceof SessionUpdateSessionUpdated) {
                    SessionUpdateSessionUpdated sessionUpdated = (SessionUpdateSessionUpdated) event;

                    // Print the full JSON representation
                    System.out.println("📄 Session Updated Event (Full JSON):");
                    String eventJson = BinaryData.fromObject(sessionUpdated).toString();
                    System.out.println(eventJson);
                }

                audioProcessor.startCapture();
            } else if (eventType == ServerEventType.INPUT_AUDIO_BUFFER_SPEECH_STARTED) {
                System.out.println("🎤 Speech detected");
                // Server handles interruption automatically with interruptResponse=true
                // Just clear any pending audio in the playback queue
                audioProcessor.skipPendingAudio();
            } else if (eventType == ServerEventType.INPUT_AUDIO_BUFFER_SPEECH_STOPPED) {
                System.out.println("🤔 Speech ended - processing...");
            } else if (eventType == ServerEventType.RESPONSE_AUDIO_DELTA) {
                // Handle audio response - extract and queue for playback
                if (event instanceof SessionUpdateResponseAudioDelta) {
                    SessionUpdateResponseAudioDelta audioEvent = (SessionUpdateResponseAudioDelta) event;
                    byte[] audioData = audioEvent.getDelta();
                    if (audioData != null && audioData.length > 0) {
                        audioProcessor.queueAudio(audioData);
                    }
                }
            } else if (eventType == ServerEventType.RESPONSE_AUDIO_DONE) {
                System.out.println("🎤 Ready for next input...");
            } else if (eventType == ServerEventType.RESPONSE_DONE) {
                System.out.println("✅ Response complete");
            } else if (eventType == ServerEventType.ERROR) {
                if (event instanceof SessionUpdateError) {
                    SessionUpdateError errorEvent = (SessionUpdateError) event;
                    System.out.println("❌ VoiceLive error: " + errorEvent.getError().getMessage());
                } else {
                    System.out.println("❌ VoiceLive error occurred");
                }
            }
        } catch (Exception e) {
            System.err.println("❌ Error handling event: " + e.getMessage());
            e.printStackTrace();
        }
    }
}

Voice Live API 开始根据模型的初始响应返回音频。 可以通过说话来中断模型。 输入“Ctrl+C”退出对话。

输出

应用程序的输出将打印到控制台。 你会看到指示系统状态的消息：

[INFO] Scanning for projects...
[INFO] 
[INFO] --------------< com.azure.ai.voicelive:model-quickstart >---------------
[INFO] Building Azure VoiceLive Model Quickstart 1.0.0
[INFO]   from pom.xml
[INFO] --------------------------------[ jar ]---------------------------------
[INFO] 
[INFO] --- exec:3.1.0:java (default-cli) @ model-quickstart ---
? Loaded configuration from application.properties
? Audio system check passed
?? Starting Voice Assistant...
   Model: gpt-realtime
   Voice: en-US-Ava:DragonHDLatestNeural
? Using API Key authentication
? Initializing VoiceLive client:
   Endpoint: https://jagoerge-voicelive-weu-resource.services.ai.azure.com/
? VoiceLive client created
? Creating session configuration:
? Session configuration created
[ModelQuickstart.main()] INFO com.azure.ai.voicelive.VoiceLiveSessionAsyncClient - WebSocket connection parameters -> endpoint: wss://my-resource.services.ai.azure.com/voice-live/realtime?api-version=2025-10-01&model=gpt-realtime headers: api-key=0XxX...x0xX
[reactor-http-nio-2] INFO com.azure.ai.voicelive.VoiceLiveSessionAsyncClient - WebSocket connection established
[reactor-http-nio-2] INFO com.azure.ai.voicelive.VoiceLiveSessionAsyncClient - Receive flux subscribed
[reactor-http-nio-2] INFO com.azure.ai.voicelive.VoiceLiveSessionAsyncClient - Send stream subscribed
[reactor-http-nio-2] INFO com.azure.ai.voicelive.VoiceLiveSessionAsyncClient - WebSocket session ready
? Session started successfully
? Subscribed to event stream
? Sending session.update configuration...
? Session configuration sent
? Audio playback started
? VOICE ASSISTANT READY
Start speaking to begin conversation
Press Ctrl+C to exit
? Session created - initializing...
? Session updated - starting microphone
? Session Updated Event (Full JSON):
{"event_id":"event_7VOMH1ALSp5A0Fa17nSZKM","session":{"model":"gpt-realtime","modalities":["audio","text"],"voice":{"name":"en-US-Ava:DragonHDLatestNeural","type":"azure-standard"},"instructions":"You are a helpful AI voice assistant. Respond naturally and conversationally. Keep your responses concise but engaging. Speak as if having a real conversation.","input_audio_sampling_rate":24000,"input_audio_format":"pcm16","output_audio_format":"pcm16","turn_detection":{"type":"server_vad","threshold":0.5,"prefix_padding_ms":300,"silence_duration_ms":500,"auto_truncate":true,"create_response":true,"interrupt_response":true},"input_audio_noise_reduction":{"type":"near_field"},"input_audio_echo_cancellation":{"type":"server_echo_cancellation"},"input_audio_transcription":{"model":"azure-speech","language":""},"tools":[],"tool_choice":"auto","temperature":0.8,"max_response_output_tokens":"inf","id":"sess_7cMSK58ShfrUY1RKnZ6Eoy"},"type":"session.updated"}
? Microphone capture started
? Audio capture loop started
? Speech detected
? Speech ended - processing...
? Ready for next input...
? Response complete
? Speech detected
? Speech ended - processing...
? Ready for next input...
? Response complete
? Speech detected
? Speech ended - processing...
? Ready for next input...
? Speech detected
? Response complete
? Speech ended - processing...

? Shutting down gracefully...
? Audio capture loop ended
? Microphone capture stopped
? Audio playback stopped

日志记录配置

此示例使用 SLF4J 进行日志记录。 默认情况下，日志记录级别设置为 INFO。 可以通过在项目根目录中创建 simplelogger.properties 文件（与 pom.xml以下文件夹相同的文件夹）来配置日志记录：

# SLF4J Simple Logger Configuration
org.slf4j.simpleLogger.defaultLogLevel=info
org.slf4j.simpleLogger.showDateTime=true
org.slf4j.simpleLogger.dateTimeFormat=yyyy-MM-dd HH:mm:ss:SSS

# Set log level for VoiceLive SDK
org.slf4j.simpleLogger.log.com.azure.ai.voicelive=debug

# Set log level for Azure Core
org.slf4j.simpleLogger.log.com.azure.core=info

若要启用调试日志记录，请将日志级别更改为 debug：

org.slf4j.simpleLogger.defaultLogLevel=debug

清理资源

完成快速入门后，可以删除创建的资源：

rm -rf voice-live-quickstart

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