Refactor settings store to rename proModel to proThinking and update related logic; enhance CSS for energy efficiency and reduced motion preferences; improve i18n translations for better clarity and consistency; modify proBlock utility functions for clearer instruction handling; streamline Vite configuration by removing unnecessary Univer.js dependencies.
This commit is contained in:
+280
-45
@@ -1,11 +1,13 @@
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import asyncio
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import base64
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import io
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import logging
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import os
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import tempfile
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import wave
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from typing import Optional
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# 设置 Hugging Face 镜像源为国内镜像
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# 设置 Hugging Face / ModelScope 镜像源为国内镜像
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os.environ.setdefault("HF_ENDPOINT", "https://hf-mirror.com")
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import numpy as np
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@@ -18,18 +20,41 @@ logger = logging.getLogger(__name__)
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# New TTS model import
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try:
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from qwen_tts import Qwen3TTSModel # type: ignore
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except Exception: # pragma: no cover
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except Exception as e: # pragma: no cover
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logger.debug("qwen_tts import failed (optional): %s", e)
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Qwen3TTSModel = None # type: ignore
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# ASR model import (MLX-based, Apple Silicon only)
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try:
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from mlx_audio.stt.models.qwen3_asr import ( # type: ignore
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ForcedAlignerModel,
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Qwen3ASRModel,
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)
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except Exception as e: # pragma: no cover
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logger.debug("mlx_audio import failed (optional): %s", e)
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Qwen3ASRModel = None # type: ignore
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ForcedAlignerModel = None # type: ignore
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try:
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from modelscope import snapshot_download # type: ignore
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except Exception as e: # pragma: no cover
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logger.debug("modelscope import failed (optional): %s", e)
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router = APIRouter()
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# Global TTS model instance
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# Global model instances
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_tts_model: Optional["Qwen3TTSModel"] = None
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_asr_model: Optional[object] = None # Qwen3ASRModel or ForcedAlignerModel
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_align_model: Optional[object] = None # Qwen3-ForcedAlignerModel
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# Model paths for loading
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MODEL_ID_HF = "Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign"
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MODEL_ID_MS = "Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign"
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# ModelScope ASR/ForcedAligner models (MLX 4-bit format)
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ASR_MODEL_ID_MS = "aufklarer/Qwen3-ASR-0.6B-MLX-4bit"
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ALIGN_MODEL_ID_MS = "aufklarer/Qwen3-ForcedAligner-0.6B-MLX"
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def _get_device_map() -> str:
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"""设备检测逻辑:优先 CUDA,其次 MPS,最后 CPU"""
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@@ -38,15 +63,14 @@ def _get_device_map() -> str:
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try:
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if hasattr(torch.backends, "mps") and torch.backends.mps.is_available():
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return "mps"
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except Exception as e:
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except Exception as e: # noqa: ANN001
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logger.debug("MPS check failed: %s", e)
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return "cpu"
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def _download_model_from_modelscope() -> Optional[str]:
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"""从 ModelScope 下载模型到本地临时目录"""
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"""从 ModelScope 下载模型到本地缓存目录"""
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try:
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from modelscope import snapshot_download
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cache_dir = os.path.join(os.path.dirname(__file__), "models")
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os.makedirs(cache_dir, exist_ok=True)
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model_dir = snapshot_download(
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@@ -56,7 +80,7 @@ def _download_model_from_modelscope() -> Optional[str]:
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)
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logger.info("ModelScope 模型下载完成: %s", model_dir)
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return model_dir
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except Exception as e:
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except Exception as e: # noqa: ANN001
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logger.warning("ModelScope 下载失败: %s", e)
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return None
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@@ -66,12 +90,22 @@ async def _warmup_tts():
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await asyncio.to_thread(_load_tts_model_with_retry)
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async def _warmup_asr():
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"""预热 ASR 模型(从 ModelScope 下载并加载)"""
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await asyncio.to_thread(_load_asr_models)
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async def _warmup_all():
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"""预热所有模型(TTS 和 ASR)"""
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logger.info("[Warmup] 开始预热 TTS 模型...")
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await _warmup_tts()
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logger.info("[Warmup] TTS 模型预热完成")
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if Qwen3ASRModel is not None:
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logger.info("[Warmup] 开始预热 ASR 模型...")
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await _warmup_asr()
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logger.info("[Warmup] ASR 模型预热完成")
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def _load_tts_model_with_retry(max_retries: int = 3) -> "Qwen3TTSModel":
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"""加载 TTS 模型,支持多个镜像源"""
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@@ -87,38 +121,136 @@ def _load_tts_model_with_retry(max_retries: int = 3) -> "Qwen3TTSModel":
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# 策略1: 尝试从 ModelScope 下载后加载
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for attempt in range(max_retries):
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try:
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logger.info("尝试从 ModelScope 下载模型...")
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logger.info("尝试从 ModelScope 下载 TTS 模型...")
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model_path = _download_model_from_modelscope()
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if model_path and os.path.isdir(model_path):
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_tts_model = Qwen3TTSModel.from_pretrained(
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_tts_model = Qwen3TTSModel.from_pretrained( # type: ignore
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model_path,
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device_map=device_map,
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dtype=torch.float16,
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)
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logger.info("ModelScope 模型加载成功: %s", model_path)
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logger.info("ModelScope TTS 模型加载成功: %s", model_path)
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return _tts_model
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except Exception as e:
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logger.warning("ModelScope 加载失败 (尝试 %d/%d): %s", attempt + 1, max_retries, e)
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except Exception as e: # noqa: ANN001
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logger.warning("ModelScope TTS 加载失败 (尝试 %d/%d): %s", attempt + 1, max_retries, e)
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last_err = e
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# 策略2: 尝试从 HuggingFace 镜像加载
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for attempt in range(max_retries):
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try:
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logger.info("尝试从 HuggingFace 镜像加载模型...")
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_tts_model = Qwen3TTSModel.from_pretrained(
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logger.info("尝试从 HuggingFace 镜像加载 TTS...")
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_tts_model = Qwen3TTSModel.from_pretrained( # type: ignore
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MODEL_ID_HF,
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device_map=device_map,
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dtype=torch.float16,
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)
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logger.info("HuggingFace 模型加载成功")
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logger.info("HuggingFace TTS 模型加载成功")
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return _tts_model
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except Exception as e:
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logger.warning("HuggingFace 加载失败 (尝试 %d/%d): %s", attempt + 1, max_retries, e)
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except Exception as e: # noqa: ANN001
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logger.warning("HuggingFace TTS 加载失败 (尝试 %d/%d): %s", attempt + 1, max_retries, e)
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last_err = e
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raise RuntimeError(f"无法加载 TTS 模型: {last_err}") from last_err
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def _load_asr_models() -> None:
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"""从 ModelScope 下载并加载 ASR/ForcedAligner MLX 模型"""
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global _asr_model, _align_model
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if snapshot_download is None:
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logger.warning("modelscope 未安装,跳过 ASR 模型加载")
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return
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if Qwen3ASRModel is None:
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logger.warning("mlx_audio 未安装,跳过 ASR 模型加载")
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return
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# Download and load ASR model from ModelScope
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try:
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logger.info("从 ModelScope 下载 ASR 模型...")
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asr_cache_dir = os.path.join(os.path.dirname(__file__), "models", "asr")
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asr_model_dir = snapshot_download(ASR_MODEL_ID_MS, cache_dir=asr_cache_dir)
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_load_asr_from_path(asr_model_dir)
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except Exception as e: # noqa: ANN001
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logger.warning("ASR ModelScope 下载失败,尝试 hf-mirror: %s", e)
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try:
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_load_asr_from_hf_mirror()
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except Exception as e2: # noqa: ANN001
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logger.warning("ASR hf-mirror 加载失败,跳过 ASR: %s", e2)
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# Download and load ForcedAligner model from ModelScope
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try:
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logger.info("从 ModelScope 下载 ForcedAligner 模型...")
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align_cache_dir = os.path.join(os.path.dirname(__file__), "models", "aligner")
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align_model_dir = snapshot_download(ALIGN_MODEL_ID_MS, cache_dir=align_cache_dir)
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_load_align_from_path(align_model_dir)
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except Exception as e: # noqa: ANN001
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logger.warning("ForcedAligner ModelScope 下载失败,尝试 hf-mirror: %s", e)
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try:
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_load_align_from_hf_mirror()
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except Exception as e2: # noqa: ANN001
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logger.warning("ForcedAligner hf-mirror 加载失败,跳过: %s", e2)
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def _load_asr_from_path(model_dir: str) -> None:
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"""从本地路径加载 ASR MLX 模型"""
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global _asr_model
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try:
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from mlx_audio.stt.utils import load as stt_load # type: ignore
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model = stt_load(model_dir)
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_asr_model = model
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logger.info("ASR 模型加载成功 (路径: %s)", model_dir)
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except Exception as e: # noqa: ANN001
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logger.warning("ASR MLX 加载失败,尝试直接构建: %s", e)
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try:
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from mlx.core import load as mx_load # type: ignore
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weights = mx_load(os.path.join(model_dir, "model.safetensors"))
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from mlx_lm import load as lm_load # type: ignore
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model = lm_load(model_dir, model_cls=Qwen3ASRModel)
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_asr_model = model
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except Exception as e2: # noqa: ANN001
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raise RuntimeError(f"无法加载 ASR MLX 模型: {e2}") from e
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def _load_asr_from_hf_mirror() -> None:
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"""从 hf-mirror 加载 ASR MLX 模型"""
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global _asr_model
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try:
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from mlx_audio.stt.utils import load as stt_load # type: ignore
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model = stt_load("mlx-community/Qwen3-ASR-0.6B-4bit")
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_asr_model = model
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except Exception as e: # noqa: ANN001
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raise RuntimeError(f"无法从 hf-mirror 加载 ASR MLX: {e}") from e
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def _load_align_from_path(model_dir: str) -> None:
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"""从本地路径加载 ForcedAligner MLX 模型"""
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global _align_model
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try:
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from mlx_audio.stt.utils import load as stt_load # type: ignore
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model = stt_load(model_dir)
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_align_model = model
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except Exception as e: # noqa: ANN001
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raise RuntimeError(f"无法加载 ForcedAligner MLX 模型 (路径: {model_dir}): {e}") from e
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def _load_align_from_hf_mirror() -> None:
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"""从 hf-mirror 加载 ForcedAligner MLX 模型"""
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global _align_model
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try:
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from mlx_audio.stt.utils import load as stt_load # type: ignore
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model = stt_load("mlx-community/Qwen3-ForcedAligner-0.6B-4bit")
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_align_model = model
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except Exception as e: # noqa: ANN001
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raise RuntimeError(f"无法从 hf-mirror 加载 ForcedAligner MLX: {e}") from e
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class TTSRequest(BaseModel):
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text: str
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instruct: str = ""
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@@ -132,28 +264,62 @@ class TTSResponse(BaseModel):
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duration_ms: int
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class ASRRequest(BaseModel):
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audio_base64: str
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language: Optional[str] = "zh-CN"
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class ASRResponse(BaseModel):
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text: str
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language: Optional[str] = None
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class ModelStatus(BaseModel):
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tts_loaded: bool
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asr_loaded: bool = False
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device: str
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tts_last_used: Optional[float] = None
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asr_last_used: Optional[float] = None
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def _ensure_model() -> "Qwen3TTSModel":
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"""确保模型已加载"""
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def _ensure_tts_model() -> "Qwen3TTSModel":
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"""确保 TTS 模型已加载"""
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global _tts_model
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if _tts_model is None:
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_tts_model = _load_tts_model_with_retry()
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return _tts_model
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def _ensure_asr_model():
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"""确保 ASR 模型已加载(懒加载)"""
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global _asr_model
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if _asr_model is None:
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try:
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from mlx_audio.stt.utils import load as stt_load # type: ignore
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_asr_model = stt_load(ASR_MODEL_ID_MS)
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except Exception as e: # noqa: ANN001
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raise RuntimeError(f"无法加载 ASR MLX 模型 (路径: {ASR_MODEL_ID_MS}): {e}") from e
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return _asr_model
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def _ensure_align_model():
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"""确保 ForcedAligner 模型已加载(懒加载)"""
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global _align_model
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if _align_model is None:
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try:
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from mlx_audio.stt.utils import load as stt_load # type: ignore
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_align_model = stt_load(ALIGN_MODEL_ID_MS)
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except Exception as e: # noqa: ANN001
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raise RuntimeError(f"无法加载 ForcedAligner MLX 模型 (路径: {ALIGN_MODEL_ID_MS}): {e}") from e
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return _align_model
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@router.get("/status", response_model=ModelStatus)
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async def get_status():
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"""获取模型状态"""
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return ModelStatus(
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tts_loaded=_tts_model is not None,
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asr_loaded=False,
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asr_loaded=_asr_model is not None,
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device=_get_device_map(),
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)
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@@ -163,13 +329,13 @@ async def get_config():
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"""获取配置信息"""
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return {
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"model": {
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"tts": "Qwen3-TTS-12Hz-1.7B-VoiceDesign",
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"asr": None,
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"tts": MODEL_ID_MS,
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"asr": ASR_MODEL_ID_MS if Qwen3ASRModel is not None else None,
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},
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"device": _get_device_map(),
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"status": {
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"tts_loaded": _tts_model is not None,
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"asr_loaded": False,
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"asr_loaded": _asr_model is not None,
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}
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}
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@@ -178,8 +344,13 @@ async def get_config():
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async def warmup_models():
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"""手动触发模型预热"""
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await _warmup_tts()
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if Qwen3ASRModel is not None:
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await _warmup_asr()
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return {
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"tts_warmup": _tts_model is not None,
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"asr_warmup": _asr_model is not None if Qwen3ASRModel else False,
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"device": _get_device_map(),
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}
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@@ -188,8 +359,8 @@ async def warmup_models():
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async def tts_endpoint(req: TTSRequest):
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"""TTS 文字转语音端点"""
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try:
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model = _ensure_model()
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except Exception as e:
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model = _ensure_tts_model()
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except Exception as e: # noqa: ANN001
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raise HTTPException(status_code=500, detail=str(e))
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text = req.text
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@@ -197,51 +368,47 @@ async def tts_endpoint(req: TTSRequest):
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try:
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# VoiceDesign 模型使用 generate_voice_design 方法
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# 返回 (wavs, sr),其中 wavs 是列表,wavs[0] 是第一个音频数据
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wavs, sr = model.generate_voice_design(
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wavs, sr = model.generate_voice_design( # type: ignore
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text=text,
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language="Chinese",
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instruct=instruct,
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)
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except Exception as e:
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except Exception as e: # noqa: ANN001
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logger.exception("TTS 推理失败")
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raise HTTPException(status_code=500, detail=f"TTS 推理失败: {e}")
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# 获取第一个音频数据
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# Get first audio data
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wav_data = wavs[0] if isinstance(wavs, (list, tuple)) else wavs
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# 转换为 numpy 数组
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if hasattr(wav_data, 'numpy'):
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wav_data = wav_data.cpu().numpy()
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# Convert to numpy array
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if hasattr(wav_data, 'numpy'): # type: ignore
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wav_data = wav_data.cpu().numpy() # type: ignore
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wav_data = np.asarray(wav_data, dtype=np.float32)
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logger.debug("wav_data shape: %s, dtype: %s, sr: %s", wav_data.shape, wav_data.dtype, sr)
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# 编码 WAV 到内存
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# Encode WAV to memory
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tmp_path = None
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||||
try:
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import soundfile as sf
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# 创建临时文件
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||||
import soundfile as sf # type: ignore
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||||
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||||
fd, tmp_path = tempfile.mkstemp(suffix=".wav")
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os.close(fd)
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os.close(fd) # type: ignore
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sf.write(tmp_path, wav_data, sr)
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||||
with open(tmp_path, "rb") as f:
|
||||
with open(tmp_path, "rb") as f: # noqa: SIM115
|
||||
audio_bytes = f.read()
|
||||
except Exception as e:
|
||||
except Exception as e: # noqa: ANN001
|
||||
logger.exception("音频编码失败")
|
||||
raise HTTPException(status_code=500, detail=f"音频编码失败: {e}")
|
||||
finally:
|
||||
# 清理临时文件
|
||||
if tmp_path and os.path.exists(tmp_path):
|
||||
if tmp_path and os.path.exists(tmp_path): # noqa: SIM201
|
||||
try:
|
||||
os.unlink(tmp_path)
|
||||
except Exception:
|
||||
except Exception as e: # noqa: ANN001
|
||||
pass
|
||||
|
||||
# 计算时长(毫秒)
|
||||
duration_ms = int(len(wav_data) / sr * 1000) if sr > 0 else 0
|
||||
|
||||
# 返回 JSON 格式,包含 base64 编码的音频
|
||||
audio_base64 = base64.b64encode(audio_bytes).decode("utf-8")
|
||||
return TTSResponse(
|
||||
audio_base64=audio_base64,
|
||||
@@ -250,6 +417,74 @@ async def tts_endpoint(req: TTSRequest):
|
||||
)
|
||||
|
||||
|
||||
@router.post("/asr", response_model=ASRResponse)
|
||||
async def asr_endpoint(req: ASRRequest):
|
||||
"""语音识别端点(非流式)"""
|
||||
if Qwen3ASRModel is None:
|
||||
raise HTTPException(status_code=501, detail="mlx_audio 未安装,ASR 功能不可用")
|
||||
|
||||
try:
|
||||
model = _ensure_asr_model()
|
||||
except Exception as e: # noqa: ANN001
|
||||
raise HTTPException(status_code=500, detail=f"ASR 模型加载失败: {e}")
|
||||
|
||||
try:
|
||||
# Decode base64 audio to WAV bytes
|
||||
audio_bytes = base64.b64decode(req.audio_base64)
|
||||
|
||||
# Load WAV file and convert to 16kHz mono numpy array
|
||||
wav_buffer = io.BytesIO(audio_bytes)
|
||||
with wave.open(wav_buffer, 'rb') as wf: # noqa: SIM115
|
||||
n_channels = wf.getnchannels()
|
||||
sampwidth = wf.getsampwidth()
|
||||
framerate = wf.getframerate()
|
||||
n_frames = wf.getnframes()
|
||||
|
||||
raw_data = wf.readframes(n_frames)
|
||||
audio_array = np.frombuffer(raw_data, dtype=np.int16 if sampwidth == 2 else np.float32)
|
||||
|
||||
# Convert to mono
|
||||
if n_channels > 1:
|
||||
audio_array = np.mean(audio_array.reshape(-1, n_channels), axis=1)
|
||||
|
||||
# Resample to 16kHz if needed
|
||||
if framerate != 16000:
|
||||
try:
|
||||
import scipy.signal as signal # type: ignore
|
||||
|
||||
n_samples = int(len(audio_array) * 16000 / framerate)
|
||||
audio_array = signal.resample(audio_array, n_samples) # type: ignore
|
||||
except Exception as e2: # noqa: ANN001
|
||||
logger.warning("重采样失败,使用原始音频: %s", e2)
|
||||
|
||||
# Convert to float32 normalized
|
||||
if audio_array.dtype == np.int16:
|
||||
audio_array = audio_array.astype(np.float32) / 32768.0
|
||||
|
||||
# Run ASR inference (non-streaming)
|
||||
result = model.generate( # type: ignore
|
||||
audio_array,
|
||||
language=req.language if req.language else None,
|
||||
)
|
||||
|
||||
# Extract text and detected language from result (STTOutput)
|
||||
recognized_text = getattr(result, 'text', str(result)) if hasattr(result, 'text') else str(result)
|
||||
detected_lang = getattr(result, 'language', req.language or "zh-CN")
|
||||
|
||||
# If language is a list (from segments), take the first one
|
||||
if isinstance(detected_lang, list) and len(detected_lang) > 0:
|
||||
detected_lang = detected_lang[0]
|
||||
|
||||
return ASRResponse(
|
||||
text=recognized_text,
|
||||
language=str(detected_lang),
|
||||
)
|
||||
|
||||
except Exception as e: # noqa: ANN001
|
||||
logger.exception("ASR 推理失败")
|
||||
raise HTTPException(status_code=500, detail=f"ASR 推理失败: {e}")
|
||||
|
||||
|
||||
def register_tts_asr_routes(app):
|
||||
"""注册 TTS/ASR 路由到 FastAPI 应用"""
|
||||
app.include_router(router, prefix="/v1/tts-asr")
|
||||
|
||||
Reference in New Issue
Block a user