asr_arena / app.py
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import gradio as gr
import torch
import torchaudio
# 定义模型路径
model_path = "https://huggingface.co/Tele-AI/TeleSpeech-ASR1.0/blob/main/large.pt"
# 下载模型
model = torch.hub.load('snakers4/silero-models', 'silero_stt', language='en', device='cpu')
torch.hub.download_url_to_file(model_path, 'large.pt')
# 加载模型
model.load_state_dict(torch.load('large.pt', map_location=torch.device('cpu')))
model.eval()
# 定义处理函数
def transcribe(audio):
waveform, sample_rate = torchaudio.load(audio.name)
waveform = waveform.unsqueeze(0)
with torch.no_grad():
output = model(waveform)
transcription = output[0]
return transcription
# 创建 Gradio 界面
iface = gr.Interface(
fn=transcribe,
inputs=gr.inputs.Audio(source="microphone", type="filepath"),
outputs="text",
title="TeleSpeech ASR",
description="Upload an audio file or record your voice to transcribe speech to text using the TeleSpeech ASR model."
)
iface.launch()