feat: 扩展后端WebSocket和语音识别功能
- 扩展websocket.py支持语音消息 - 优化asr_service.py语音识别服务 - 更新main.py和requirements.txt - 更新.env.production配置 Co-authored-by: Cursor <cursoragent@cursor.com>
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@@ -46,4 +46,14 @@ TENCENT_SMS_TEMPLATE_ID=2592163
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# 短信模板参数数量(1=仅验证码,2=验证码+过期时间)
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# 如果遇到 TemplateParamSetNotMatchApprovedTemplate 错误,请检查腾讯云控制台中的模板配置
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# 并根据实际模板参数数量设置此值
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TENCENT_SMS_TEMPLATE_PARAM_COUNT=1
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TENCENT_SMS_TEMPLATE_PARAM_COUNT=1
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# CPU 环境(推荐 small + int8)
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ASR_MODEL_SIZE=small
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ASR_DEVICE=cpu
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ASR_COMPUTE_TYPE=int8
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# GPU 环境(推荐 medium + float16)
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# ASR_MODEL_SIZE=medium
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# ASR_DEVICE=cuda
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# ASR_COMPUTE_TYPE=float16
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12
api/main.py
12
api/main.py
@@ -94,6 +94,7 @@ app = FastAPI(title="Life Echo API", version="1.0.0")
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@app.on_event("startup")
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async def startup_event():
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"""应用启动事件"""
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import asyncio
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logger.info("=" * 50)
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logger.info("Life Echo API 正在启动...")
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logger.info("=" * 50)
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@@ -105,6 +106,17 @@ async def startup_event():
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logger.info("Redis 连接已建立")
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except Exception as e:
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logger.warning(f"Redis 连接失败(会话存储将不可用): {e}")
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# 检查并预加载 ASR 模型(在后台线程执行,避免阻塞启动)
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try:
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from services.asr_service import asr_service
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asr_ready = await asyncio.to_thread(asr_service.ensure_ready)
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if asr_ready:
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logger.info("ASR 模型已就绪(本地 Whisper)")
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else:
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logger.warning("ASR 模型未就绪,语音转写将不可用")
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except Exception as e:
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logger.warning(f"ASR 初始化检查失败: {e}")
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@app.on_event("shutdown")
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@@ -38,9 +38,8 @@ httpx==0.27.0
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python-jose[cryptography]==3.3.0
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bcrypt>=4.0.0
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# Audio Processing (optional, for future ASR/TTS integration)
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# pydub==0.25.1
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# speech-recognition==3.10.4
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# Audio Processing - Local Whisper ASR
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faster-whisper>=1.0.0
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# Image Processing
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Pillow>=10.0.0
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@@ -19,6 +19,7 @@ from database.models import Conversation, Segment
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from database.models import User as UserModel
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from services.auth_service import verify_token
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from services.memoir_state_service import get_or_create_state
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from services.asr_service import asr_service
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from fastapi import HTTPException, status
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logger = logging.getLogger(__name__)
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@@ -28,8 +29,9 @@ class MessageType(str, Enum):
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"""WebSocket 消息类型"""
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CONNECT = "connect"
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AUDIO_CHUNK = "audio_chunk"
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AUDIO_MESSAGE = "audio_message" # 完整音频消息(类似微信语音)
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TEXT = "text" # 文本消息
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TRANSCRIPT = "transcript"
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TRANSCRIPT = "transcript" # 语音转文字结果
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AGENT_RESPONSE = "agent_response"
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TTS_AUDIO = "tts_audio"
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END_CONVERSATION = "end_conversation"
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@@ -190,6 +192,70 @@ async def websocket_endpoint(
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manager=manager
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)
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elif msg_type == MessageType.AUDIO_MESSAGE:
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# 处理完整音频消息(类似微信语音)
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data = message.get("data", {})
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audio_base64 = data.get("audio_base64", "")
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audio_duration = data.get("duration", 0)
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if audio_base64:
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logger.info(f"收到音频消息,时长: {audio_duration}s")
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try:
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# 1. ASR 转写
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transcript_text = await asr_service.transcribe(audio_base64)
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logger.info(f"ASR 转写结果: {transcript_text}")
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# 2. 发送转写结果给客户端
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await manager.send_message(conversation_id, {
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"type": MessageType.TRANSCRIPT,
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"conversation_id": conversation_id,
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"data": {
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"text": transcript_text,
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"audio_duration": audio_duration
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},
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"timestamp": datetime.now(timezone.utc).isoformat()
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})
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# 3. 保存段落到数据库(包含转写文本和音频信息)
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segment = Segment(
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id=str(uuid.uuid4()),
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conversation_id=conversation_id,
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transcript_text=transcript_text,
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audio_url=f"audio:{audio_duration}s", # 简化存储,标记为音频消息
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processed=False
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)
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db.add(segment)
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await db.commit()
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await db.refresh(segment)
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await manager.background_runner.queue_message(conversation.user_id, segment.id)
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# 4. Agent 生成回应(基于转写文本)
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if transcript_text and not transcript_text.startswith("转写失败"):
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await process_user_message(
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conversation_id=conversation_id,
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user_message=transcript_text,
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conversation=conversation,
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segment=segment,
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db=db,
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manager=manager
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)
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else:
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# 转写失败,发送错误消息
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await manager.send_message(conversation_id, {
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"type": MessageType.ERROR,
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"data": {"message": "语音转写失败,请重试或使用文字输入"},
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"timestamp": datetime.now(timezone.utc).isoformat()
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})
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except Exception as e:
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logger.error(f"处理音频消息失败: {e}", exc_info=True)
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await manager.send_message(conversation_id, {
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"type": MessageType.ERROR,
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"data": {"message": f"处理音频消息失败: {str(e)}"},
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"timestamp": datetime.now(timezone.utc).isoformat()
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})
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elif msg_type == MessageType.END_CONVERSATION:
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# 结束对话
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conversation.status = "ended"
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@@ -1,23 +1,95 @@
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"""
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ASR 服务:语音转文字
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使用本地 faster-whisper 模型进行语音识别
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"""
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import base64
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import logging
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import os
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import tempfile
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from typing import Optional
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from openai import OpenAI
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logger = logging.getLogger(__name__)
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# 模型配置
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# 可选模型: tiny, base, small, medium, large-v2, large-v3
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# tiny/base 适合 CPU,small/medium 需要更多资源,large 需要 GPU
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ASR_MODEL_SIZE = os.getenv("ASR_MODEL_SIZE", "small")
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ASR_DEVICE = os.getenv("ASR_DEVICE", "auto") # auto, cpu, cuda
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ASR_COMPUTE_TYPE = os.getenv("ASR_COMPUTE_TYPE", "auto") # auto, int8, float16, float32
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class ASRService:
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"""ASR 服务(语音转文字)"""
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"""
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ASR 服务(语音转文字)
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使用 faster-whisper 本地模型
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"""
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def __init__(self):
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api_key = os.getenv("OPENAI_API_KEY", "")
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if api_key:
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self.client = OpenAI(api_key=api_key)
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else:
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self.client = None
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self.model = None
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self._model_loaded = False
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self._load_error = None
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async def transcribe(self, audio_base64: str) -> str | None:
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def _load_model(self) -> bool:
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"""加载模型(首次调用时执行,后续直接返回)。返回是否加载成功。"""
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if self._model_loaded:
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return self.model is not None
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try:
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from faster_whisper import WhisperModel
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logger.info(f"正在加载 Whisper 模型: {ASR_MODEL_SIZE}, device={ASR_DEVICE}, compute_type={ASR_COMPUTE_TYPE}")
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# 确定设备和计算类型
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device = ASR_DEVICE
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compute_type = ASR_COMPUTE_TYPE
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if device == "auto":
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# 自动检测:优先使用 CUDA,否则使用 CPU
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try:
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import torch
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device = "cuda" if torch.cuda.is_available() else "cpu"
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except ImportError:
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device = "cpu"
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if compute_type == "auto":
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# 根据设备自动选择计算类型
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if device == "cuda":
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compute_type = "float16" # GPU 使用 float16
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else:
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compute_type = "int8" # CPU 使用 int8 量化,速度更快
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self.model = WhisperModel(
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ASR_MODEL_SIZE,
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device=device,
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compute_type=compute_type
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)
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self._model_loaded = True
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logger.info(f"Whisper 模型加载成功: {ASR_MODEL_SIZE} on {device} ({compute_type})")
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return True
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except ImportError as e:
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self._load_error = "faster-whisper 未安装,请运行: pip install faster-whisper"
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logger.error(self._load_error)
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return False
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except Exception as e:
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self._load_error = f"加载 Whisper 模型失败: {str(e)}"
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logger.error(self._load_error, exc_info=True)
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return False
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def ensure_ready(self) -> bool:
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"""
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确保 ASR 模型已就绪(用于启动时预加载与检查)。
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可在应用初始化时调用;为同步阻塞调用,建议在后台线程执行。
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返回是否就绪。
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"""
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return self._load_model()
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def is_ready(self) -> bool:
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"""检查 ASR 模型是否已加载并可用。"""
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return self._model_loaded and self.model is not None
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async def transcribe(self, audio_base64: str) -> Optional[str]:
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"""
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转写音频为文字
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@@ -25,39 +97,56 @@ class ASRService:
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audio_base64: Base64 编码的音频数据
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Returns:
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转写文本
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转写文本,失败时返回错误信息
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"""
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if not self.client:
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# 如果没有配置 API Key,返回模拟数据
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return "这是模拟的转写文本(请配置 OPENAI_API_KEY 以使用实际 ASR 功能)"
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# 懒加载模型
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self._load_model()
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if not self.model:
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error_msg = self._load_error or "ASR 模型未加载"
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logger.warning(error_msg)
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return f"转写失败: {error_msg}"
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tmp_file_path = None
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try:
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# 解码 Base64 音频
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audio_bytes = base64.b64decode(audio_base64)
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# 保存临时文件
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import tempfile
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with tempfile.NamedTemporaryFile(suffix=".m4a", delete=False) as tmp_file:
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tmp_file.write(audio_bytes)
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tmp_file_path = tmp_file.name
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try:
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# 调用 OpenAI Whisper API
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with open(tmp_file_path, "rb") as audio_file:
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transcript = self.client.audio.transcriptions.create(
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model="whisper-1",
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file=audio_file,
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language="zh" # 中文
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)
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return transcript.text
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finally:
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# 清理临时文件
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import os
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if os.path.exists(tmp_file_path):
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os.remove(tmp_file_path)
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# 使用 faster-whisper 转写
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# language="zh" 指定中文,可以提高识别速度
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# beam_size=5 是默认值,可以调整
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segments, info = self.model.transcribe(
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tmp_file_path,
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language="zh",
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beam_size=5,
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vad_filter=True, # 启用 VAD 过滤静音部分
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vad_parameters=dict(
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min_silence_duration_ms=500, # 最小静音时长
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)
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)
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# 合并所有转写片段
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transcript_text = "".join(segment.text for segment in segments)
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logger.info(f"ASR 转写完成: 语言={info.language}, 概率={info.language_probability:.2f}, 文本长度={len(transcript_text)}")
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return transcript_text.strip() if transcript_text else ""
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except Exception as e:
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# 出错时返回错误信息
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logger.error(f"ASR 转写失败: {e}", exc_info=True)
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return f"转写失败: {str(e)}"
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finally:
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# 清理临时文件
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if tmp_file_path and os.path.exists(tmp_file_path):
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try:
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os.remove(tmp_file_path)
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except Exception:
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pass
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# 全局实例
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