Update app.py
Browse files
app.py
CHANGED
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@@ -1,7 +1,15 @@
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import streamlit as st
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import subprocess
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import json
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import os
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import requests
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import torch
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import whisperx
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@@ -92,7 +100,8 @@ with st.sidebar:
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fps = st.number_input("Timeline FPS", value=25)
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st.header("Model Settings")
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num_speakers = st.number_input("Speakers (0=Auto)", min_value=0, value=0)
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st.divider()
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@@ -111,7 +120,10 @@ if uploaded_file:
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if not ACTIVE_HF_TOKEN or "PASTE_YOUR_HF_TOKEN" in ACTIVE_HF_TOKEN:
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st.error("Please provide a valid Hugging Face Token.")
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else:
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-
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# Save local temp file
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with open("temp_input", "wb") as f:
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f.write(uploaded_file.getbuffer())
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@@ -126,10 +138,12 @@ if uploaded_file:
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try:
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device = "cuda" if torch.cuda.is_available() else "cpu"
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st.write(f"🚀 **Loading WhisperX on {device}...**")
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# 1. Transcribe
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batch_size = 16 # Reduce if low VRAM
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# Use float16 for GPU, int8 for CPU
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compute_type = "float16" if device == "cuda" else "int8"
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@@ -137,14 +151,14 @@ if uploaded_file:
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st.write("📝 **Transcribing...**")
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audio = whisperx.load_audio("temp_audio.wav")
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result = model.transcribe(audio, batch_size=
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# Cleanup VRAM
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gc.collect()
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torch.cuda.empty_cache()
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del model
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# 2. Align (Improves timestamp accuracy)
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st.write("⏱️ **Aligning Audio...**")
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model_a, metadata = whisperx.load_align_model(language_code=result["language"], device=device)
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result = whisperx.align(result["segments"], model_a, metadata, audio, device, return_char_alignments=False)
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@@ -170,10 +184,11 @@ if uploaded_file:
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# Format for Gemini
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processed_segments = []
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for segment in final_result["segments"]:
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processed_segments.append({
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"speaker": segment.get("speaker", "Unknown"),
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"text": segment["text"],
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"start": segment["start"],
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"end": segment["end"]
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})
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import os
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import numpy as np
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# --- NUMPY 2.0 PATCH ---
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# This must run before any other library imports to prevent crashes
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# with pyannote/whisperx which might expect the old 'np.NaN' attribute.
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if not hasattr(np, 'NaN'):
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np.NaN = np.nan
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import streamlit as st
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import subprocess
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import json
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import requests
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import torch
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import whisperx
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fps = st.number_input("Timeline FPS", value=25)
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st.header("Model Settings")
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# T4 has 16GB VRAM, large-v2 works well
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model_size = st.selectbox("Whisper Model", ["large-v2", "medium"], index=0)
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num_speakers = st.number_input("Speakers (0=Auto)", min_value=0, value=0)
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st.divider()
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if not ACTIVE_HF_TOKEN or "PASTE_YOUR_HF_TOKEN" in ACTIVE_HF_TOKEN:
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st.error("Please provide a valid Hugging Face Token.")
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else:
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status_container = st.empty()
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with status_container.container():
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st.write("🔄 **Processing Started...**")
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# Save local temp file
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with open("temp_input", "wb") as f:
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f.write(uploaded_file.getbuffer())
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try:
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device = "cuda" if torch.cuda.is_available() else "cpu"
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if device == "cpu":
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st.warning("⚠️ No GPU detected. WhisperX will be very slow.")
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st.write(f"🚀 **Loading WhisperX on {device}...**")
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# 1. Transcribe
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# Use float16 for GPU, int8 for CPU
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compute_type = "float16" if device == "cuda" else "int8"
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st.write("📝 **Transcribing...**")
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audio = whisperx.load_audio("temp_audio.wav")
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result = model.transcribe(audio, batch_size=16)
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# Cleanup VRAM
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gc.collect()
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torch.cuda.empty_cache()
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del model
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# 2. Align (Improves timestamp accuracy for diarization)
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st.write("⏱️ **Aligning Audio...**")
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model_a, metadata = whisperx.load_align_model(language_code=result["language"], device=device)
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result = whisperx.align(result["segments"], model_a, metadata, audio, device, return_char_alignments=False)
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# Format for Gemini
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processed_segments = []
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# WhisperX structure is slightly different, it returns 'segments' list
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for segment in final_result["segments"]:
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processed_segments.append({
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"speaker": segment.get("speaker", "Unknown"),
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"text": segment["text"].strip(),
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"start": segment["start"],
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"end": segment["end"]
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})
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