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import gradio as gr | |
from transformers import pipeline, AutoTokenizer, AutoModelForCausalLM | |
import numpy as np | |
import torch | |
transcriber = pipeline("automatic-speech-recognition", model="openai/whisper-base.en") | |
# Load a GPT-2 model for general question answering | |
tokenizer = AutoTokenizer.from_pretrained("gpt2-medium", cache_dir="./cache") | |
model = AutoModelForCausalLM.from_pretrained("gpt2-medium", cache_dir="./cache") | |
def transcribe(audio): | |
if audio is None: | |
return "No audio recorded." | |
sr, y = audio | |
y = y.astype(np.float32) | |
y /= np.max(np.abs(y)) | |
return transcriber({"sampling_rate": sr, "raw": y})["text"] | |
def answer(question): | |
input_ids = tokenizer.encode(f"Q: {question}\nA:", return_tensors="pt") | |
# Generate a response | |
with torch.no_grad(): | |
output = model.generate(input_ids, max_length=150, num_return_sequences=1, | |
temperature=0.7, top_k=50, top_p=0.95) | |
response = tokenizer.decode(output[0], skip_special_tokens=True) | |
# Extract only the answer part | |
answer = response.split("A:")[-1].strip() | |
print(answer) | |
return response | |
def process_audio(audio): | |
if audio is None: | |
return "No audio recorded.", "" | |
transcription = transcribe(audio) | |
answer_result = answer(transcription) | |
return transcription, answer_result | |
def clear_all(): | |
return None, "", "" | |
with gr.Blocks() as demo: | |
gr.Markdown("# Audio Transcription and Question Answering") | |
audio_input = gr.Audio(label="Audio Input", sources=["microphone"], type="numpy") | |
transcription_output = gr.Textbox(label="Transcription") | |
answer_output = gr.Textbox(label="Answer Result", lines=10) | |
clear_button = gr.Button("Clear") | |
audio_input.stop_recording( | |
fn=process_audio, | |
inputs=[audio_input], | |
outputs=[transcription_output, answer_output] | |
) | |
clear_button.click( | |
fn=clear_all, | |
inputs=[], | |
outputs=[audio_input, transcription_output, answer_output] | |
) | |
demo.launch() |