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import gradio as gr | |
from transformers import pipeline | |
import edge_tts | |
import numpy as np | |
import asyncio | |
import os | |
# Load STT and chatbot pipelines | |
stt = pipeline("automatic-speech-recognition", model="openai/whisper-small") | |
chatbot = pipeline("text-generation", model="HooshvareLab/gpt2-fa") | |
async def tts(text: str, voice: str = "fa-IR-FaridNeural"): | |
communicate = edge_tts.Communicate(text, voice) | |
audio_data = b"" | |
async for chunk in communicate.stream(): | |
if chunk["type"] == "audio": | |
audio_data += chunk["data"] | |
audio_array = np.frombuffer(audio_data, dtype=np.int16) | |
sample_rate = 24000 | |
return sample_rate, audio_array | |
async def audio_to_audio(audio_input): | |
if audio_input is None: | |
return None, "No audio input received." | |
sample_rate_in, data_in = audio_input | |
audio = {"array": data_in, "sampling_rate": sample_rate_in} | |
# 1. ASR → text | |
text = stt(audio)["text"] | |
# 2. Generate response | |
response = chatbot(text, max_length=50, num_return_sequences=1)[0]["generated_text"] | |
# 3. TTS | |
return await tts(response) | |
# Gradio interface | |
demo = gr.Interface( | |
fn=audio_to_audio, | |
inputs=gr.Audio( | |
sources=["microphone"], # Use 'sources' instead of deprecated 'source' :contentReference[oaicite:2]{index=2} | |
type="numpy", | |
label="Speak in Farsi" | |
), | |
outputs=gr.Audio(type="numpy", label="Response in Farsi"), | |
title="Farsi Audio Chatbot", | |
description="Speak in Farsi, and the app will respond in Farsi audio.", | |
allow_flagging="never" | |
) | |
if __name__ == "__main__": | |
demo.launch( | |
server_name="0.0.0.0", | |
server_port=int(os.environ.get("PORT", 7860)) | |
) | |