mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-21 22:53:56 +08:00
+4
-1
@@ -141,8 +141,8 @@ music/
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# Cython debug symbols
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cython_debug/
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*.iml
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model.pt
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tmp
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.history
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.DS_Store
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main/xiaozhi-server/data
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main/manager-web/node_modules
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@@ -151,3 +151,6 @@ main/manager-web/node_modules
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.private_config.yaml
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.env.development
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# model files
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main/xiaozhi-server/models/SenseVoiceSmall/model.pt
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main/xiaozhi-server/models/sherpa-onnx*
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@@ -208,6 +208,7 @@ server:
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| 类型 | 平台名称 | 使用方式 | 收费模式 | 备注 |
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|:---:|:---------:|:----:|:----:|:--:|
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| ASR | FunASR | 本地使用 | 免费 | |
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| ASR | SherpaASR | 本地使用 | 免费 | |
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| ASR | DoubaoASR | 接口调用 | 收费 | |
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---
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@@ -165,6 +165,7 @@ In fact, any LLM that supports OpenAI API calls can be integrated.
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| Type | Platform Name | Usage Method | Pricing Model | Remarks |
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|:----:|:-------------------:|:------------:|:-------------:|:-------:|
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| ASR | FunASR | Local | Free | |
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| ASR | SherpaASR | Local | Free | |
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| ASR | DoubaoASR | API call | Paid | |
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---
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@@ -120,6 +120,10 @@ ASR:
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type: fun_local
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model_dir: models/SenseVoiceSmall
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output_dir: tmp/
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SherpaASR:
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type: sherpa_onnx_local
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model_dir: models/sherpa-onnx-sense-voice-zh-en-ja-ko-yue-2024-07-17
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output_dir: tmp/
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DoubaoASR:
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type: doubao
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appid: 你的火山引擎语音合成服务appid
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@@ -0,0 +1,164 @@
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import time
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import wave
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import os
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import sys
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import io
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from config.logger import setup_logging
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from typing import Optional, Tuple, List
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import uuid
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import opuslib_next
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from core.providers.asr.base import ASRProviderBase
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import numpy as np
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import sherpa_onnx
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from modelscope.hub.file_download import model_file_download
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TAG = __name__
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logger = setup_logging()
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# 捕获标准输出
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class CaptureOutput:
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def __enter__(self):
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self._output = io.StringIO()
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self._original_stdout = sys.stdout
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sys.stdout = self._output
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def __exit__(self, exc_type, exc_value, traceback):
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sys.stdout = self._original_stdout
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self.output = self._output.getvalue()
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self._output.close()
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# 将捕获到的内容通过 logger 输出
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if self.output:
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logger.bind(tag=TAG).info(self.output.strip())
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class ASRProvider(ASRProviderBase):
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def __init__(self, config: dict, delete_audio_file: bool):
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self.model_dir = config.get("model_dir")
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self.output_dir = config.get("output_dir")
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self.delete_audio_file = delete_audio_file
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# 确保输出目录存在
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os.makedirs(self.output_dir, exist_ok=True)
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# 初始化模型文件路径
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model_files = {
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"model.int8.onnx": os.path.join(self.model_dir, "model.int8.onnx"),
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"tokens.txt": os.path.join(self.model_dir, "tokens.txt")
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}
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# 下载并检查模型文件
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try:
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for file_name, file_path in model_files.items():
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if not os.path.isfile(file_path):
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logger.bind(tag=TAG).info(f"正在下载模型文件: {file_name}")
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model_file_download(
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model_id="pengzhendong/sherpa-onnx-sense-voice-zh-en-ja-ko-yue",
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file_path=file_name,
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local_dir=self.model_dir
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)
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if not os.path.isfile(file_path):
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raise FileNotFoundError(f"模型文件下载失败: {file_path}")
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self.model_path = model_files["model.int8.onnx"]
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self.tokens_path = model_files["tokens.txt"]
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except Exception as e:
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logger.bind(tag=TAG).error(f"模型文件处理失败: {str(e)}")
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raise
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with CaptureOutput():
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self.model = sherpa_onnx.OfflineRecognizer.from_sense_voice(
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model=self.model_path,
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tokens=self.tokens_path,
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num_threads=2,
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sample_rate=16000,
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feature_dim=80,
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decoding_method="greedy_search",
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debug=False,
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use_itn=True,
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)
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def save_audio_to_file(self, opus_data: List[bytes], session_id: str) -> str:
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"""将Opus音频数据解码并保存为WAV文件"""
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file_name = f"asr_{session_id}_{uuid.uuid4()}.wav"
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file_path = os.path.join(self.output_dir, file_name)
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decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
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pcm_data = []
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for opus_packet in opus_data:
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try:
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pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
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pcm_data.append(pcm_frame)
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except opuslib_next.OpusError as e:
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logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
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with wave.open(file_path, "wb") as wf:
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wf.setnchannels(1)
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wf.setsampwidth(2) # 2 bytes = 16-bit
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wf.setframerate(16000)
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wf.writeframes(b"".join(pcm_data))
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return file_path
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def read_wave(self, wave_filename: str) -> Tuple[np.ndarray, int]:
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"""
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Args:
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wave_filename:
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Path to a wave file. It should be single channel and each sample should
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be 16-bit. Its sample rate does not need to be 16kHz.
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Returns:
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Return a tuple containing:
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- A 1-D array of dtype np.float32 containing the samples, which are
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normalized to the range [-1, 1].
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- sample rate of the wave file
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"""
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with wave.open(wave_filename) as f:
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assert f.getnchannels() == 1, f.getnchannels()
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assert f.getsampwidth() == 2, f.getsampwidth() # it is in bytes
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num_samples = f.getnframes()
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samples = f.readframes(num_samples)
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samples_int16 = np.frombuffer(samples, dtype=np.int16)
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samples_float32 = samples_int16.astype(np.float32)
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samples_float32 = samples_float32 / 32768
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return samples_float32, f.getframerate()
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async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
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"""语音转文本主处理逻辑"""
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file_path = None
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try:
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# 保存音频文件
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start_time = time.time()
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file_path = self.save_audio_to_file(opus_data, session_id)
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logger.bind(tag=TAG).debug(f"音频文件保存耗时: {time.time() - start_time:.3f}s | 路径: {file_path}")
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# 语音识别
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start_time = time.time()
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s = self.model.create_stream()
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samples, sample_rate = self.read_wave(file_path)
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s.accept_waveform(sample_rate, samples)
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self.model.decode_stream(s)
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text = s.result.text
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logger.bind(tag=TAG).debug(f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}")
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return text, file_path
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except Exception as e:
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logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
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return "", None
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finally:
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# 文件清理逻辑
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if self.delete_audio_file and file_path and os.path.exists(file_path):
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try:
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os.remove(file_path)
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logger.bind(tag=TAG).debug(f"已删除临时音频文件: {file_path}")
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except Exception as e:
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logger.bind(tag=TAG).error(f"文件删除失败: {file_path} | 错误: {e}")
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@@ -19,4 +19,6 @@ loguru==0.7.3
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requests==2.32.3
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cozepy==0.12.0
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mem0ai==0.1.62
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bs4==0.0.2
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bs4==0.0.2
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modelscope==1.23.2
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sherpa_onnx==1.11.0
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