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Update app.py
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app.py
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# app.py
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import torch
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
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import gradio as gr
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import os
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# --- 1.
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device = "cpu"
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torch_dtype = torch.float32 # Use float32 for CPU
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print("--- Loading model and processor ---")
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# Load the model and processor
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model = AutoModelForSpeechSeq2Seq.from_pretrained(
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model_id,
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torch_dtype=torch_dtype,
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low_cpu_mem_usage=True,
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use_safetensors=True # Safetensors is generally preferred
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)
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processor = AutoProcessor.from_pretrained(model_id)
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# Create the pipeline
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print("--- Creating transcription pipeline ---")
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pipe = pipeline(
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"automatic-speech-recognition",
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model=model,
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tokenizer=processor.tokenizer,
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feature_extractor=processor.feature_extractor,
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max_new_tokens=128,
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torch_dtype=torch_dtype,
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device=device,
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)
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""
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# app.py
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import streamlit as st
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import torch
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
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import os
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# --- 1. تنظیمات اولیه و عنوان صفحه ---
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st.set_page_config(
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page_title="Persian Whisper ASR",
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page_icon="🇮🇷🎙️",
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layout="centered"
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)
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st.title("🇮🇷 اپلیکیشن تبدیل گفتار به نوشتار فارسی (Whisper)")
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st.markdown("""
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این یک نسخه نمایشی برای مدل **`vhdm/whisper-large-fa-v1`** است.
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فایل صوتی خود را آپلود کنید تا متن آن را مشاهده نمایید.
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""")
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# --- 2. بارگذاری مدل (با کش کردن برای سرعت بیشتر) ---
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# این دکوراتور به Streamlit میگوید که مدل را فقط یک بار بارگذاری کند.
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@st.cache_resource
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def load_model():
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"""Loads and caches the Whisper model and processor."""
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print("--- Loading model and processor for the first time ---")
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device = "cpu"
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torch_dtype = torch.float32
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model_id = "vhdm/whisper-large-fa-v1"
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model = AutoModelForSpeechSeq2Seq.from_pretrained(
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model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
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)
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processor = AutoProcessor.from_pretrained(model_id)
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pipe = pipeline(
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"automatic-speech-recognition",
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model=model,
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tokenizer=processor.tokenizer,
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feature_extractor=processor.feature_extractor,
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max_new_tokens=128,
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torch_dtype=torch_dtype,
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device=device,
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)
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print("--- Model loaded successfully ---")
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return pipe
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# مدل را بارگذاری میکنیم
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transcription_pipe = load_model()
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# --- 3. بخش آپلود فایل و پردازش ---
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st.header("فایل صوتی خود را آپلود کنید")
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uploaded_file = st.file_uploader(
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"یک فایل صوتی انتخاب کنید (WAV, MP3, FLAC)...",
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type=["wav", "mp3", "m4a", "flac"]
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if uploaded_file is not None:
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# نمایش فایل صوتی
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st.audio(uploaded_file, format='audio/wav')
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# دکمه برای شروع پردازش
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if st.button("شروع رونویسی"):
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# برای پردازش، فایل را به صورت موقت ذخیره میکنیم
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temp_file_path = f"./temp_{uploaded_file.name}"
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with open(temp_file_path, "wb") as f:
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f.write(uploaded_file.getbuffer())
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# نمایش پیام در حال پردازش
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with st.spinner("در حال پردازش فایل صوتی... لطفاً صبر کنید."):
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result = transcription_pipe(temp_file_path)
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transcription = result["text"]
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# نمایش نتیجه
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st.success("پردازش با موفقیت انجام شد!")
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st.subheader("متن رونویسی شده:")
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st.write(transcription)
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# حذف فایل موقت
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os.remove(temp_file_path)
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