app.py

import os
import gradio as gr
import whisper
import librosa
import torch
from transformers import Wav2Vec2Processor, Wav2Vec2Tokenizer

device = "cuda" if torch.cuda.is_available() else "cpu"

def audio_to_text(audio):
    model = whisper.load_model("base")

    audio = whisper.load_audio(audio)
    result = model.transcribe(audio)

    return result["text"]
    # tokenizer = Wav2Vec2Tokenizer.from_pretrained("facebook/wav2vec2-base-960h")

    # logits = preprocess(audio)

    # predicted_ids = torch.argmax(logits, dim=-1)
    # transcriptions = tokenizer.decode(predicted_ids[0])
    # return transcriptions

def preprocess(audio):
    model_save_path = "model_save"
    model_name = "wav2vec2_osr_version_1"
    speech, rate = librosa.load(audio, sr=16000)
    model_path = os.path.join(model_save_path, model_name+".pt")
    pipeline_path = os.path.join(model_save_path, model_name+"_vocab")

    access_token = "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"
    processor = Wav2Vec2Processor.from_pretrained(pipeline_path, use_auth_token=access_token)
    model = torch.load(model_path)
    model.eval()
    input_values = processor(speech, sampling_rate=rate, return_tensors="pt").input_values.to(device)
    logits = model(input_values).logits
    return logits

demo = gr.Interface(
    fn=audio_to_text,
    inputs=gr.Audio(source="upload", type="filepath"),
    examples=[["example.flac"]],
    outputs="text"
)
demo.launch()