import gradio as gr import torch from wenet.cli.model import load_model import os from huggingface_hub import login # Load the API token from the environment variables api_token = os.getenv('HUGGINGFACE_API_TOKEN') if not api_token: raise ValueError("No Hugging Face API token found. Please set the HUGGING_FACE_API_TOKEN environment variable.") # Login to Hugging Face Hub login(token=api_token, add_to_git_credential=True) def process_cat_embs(cat_embs): device = "cpu" cat_embs = torch.tensor( [float(c) for c in cat_embs.split(',')]).to(device) return cat_embs def download_rev_models(): from huggingface_hub import hf_hub_download import joblib REPO_ID = "Revai/reverb-asr" files = ['reverb_asr_v1.jit.zip', 'tk.units.txt'] downloaded_files = [hf_hub_download(repo_id=REPO_ID, filename=f) for f in files] model = load_model(downloaded_files[0], downloaded_files[1]) return model model = download_rev_models() def recognition(audio, style=0): if audio is None: return "Input Error! Please enter one audio!" cat_embs = ','.join([str(s) for s in (style, 1-style)]) cat_embs = process_cat_embs(cat_embs) ans = model.transcribe(audio, cat_embs = cat_embs) if ans is None: return "ERROR! No text output! Please try again!" txt = ans['text'] txt = txt.replace('▁', ' ') return txt audio_input = gr.Audio(type="filepath", label="Upload or Record Audio") style_slider = gr.Slider(0, 1, value=0, step=0.1, label="Transcription Style", info="Adjust the transcription style: 0 (casual) to 1 (formal).") output_textbox = gr.Textbox(label="Transcription Output") text = "ASR Transcription Opensource Demo-CPU" # description description = ( " Opensource Automatic Speech Recognition in English" "Verbatim Transcript style(1) refers to word to word-to-word transcription of an audio" "Non Verbatim Transcript style(0) refers to just conserving the message of the original audio" ) iface = gr.Interface( fn=recognition, inputs=[audio_input, style_slider], outputs=output_textbox, title=text, description=description, theme='default', ) iface.launch()