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!pip install -U git+https://github.com/openai/whisper.git
!pip install -U gradio
!pip install torch
!pip install numpy
!pip install ffmpeg-python
!pip install tqdm
!pip install jiwer
from huggingface_hub import InferenceClient
import os
import asyncio
import whisper
import gradio as gr
import torch
import shutil
import logging
from pathlib import Path
import concurrent.futures
import ffmpeg
import re
import threading
from tqdm.notebook import tqdm
import numpy as np
# --- File Handling ---
# Define paths and constants
TEMP_FOLDER = '/content/temp/'
SUPPORTED_AUDIO_FORMATS = ['.mp3', '.wav', '.aac', '.flac', '.ogg', '.m4a', '.amr', '.wma']
SUPPORTED_VIDEO_FORMATS = ['.mp4', '.avi', '.mov', '.wmv', '.mkv', '.webm', '.3gp']
SUPPORTED_FORMATS = SUPPORTED_AUDIO_FORMATS + SUPPORTED_VIDEO_FORMATS
def create_folders():
"""Creates the necessary temporary folder if it doesn't exist."""
Path(TEMP_FOLDER).mkdir(parents=True, exist_ok=True)
def is_supported_format(file):
"""Checks if a file has a supported audio/video format."""
if file is not None:
return any(file.lower().endswith(ext) for ext in SUPPORTED_FORMATS)
else:
return False
def convert_to_wav(original_file_path):
"""Converts input file to WAV format."""
output_path = os.path.join(TEMP_FOLDER, os.path.splitext(os.path.basename(original_file_path))[0] + '.wav')
try:
(
ffmpeg
.input(original_file_path)
.output(output_path, acodec='pcm_s16le', ac=1, ar='16k')
.overwrite_output()
.run(capture_stdout=True, capture_stderr=True)
)
return output_path
except ffmpeg.Error as e:
print(f'Error converting {original_file_path}: {e.stderr.decode()}')
return None
def delete_temp_file(file_path):
"""Deletes a temporary file."""
if os.path.exists(file_path):
os.remove(file_path)
# --- Transcription ---
class WhisperModelCache:
"""Singleton class to load and cache the Whisper model."""
_instance = None
@staticmethod
def get_instance():
"""Get the singleton instance."""
if WhisperModelCache._instance is None:
WhisperModelCache._instance = WhisperModelCache()
return WhisperModelCache._instance
def __init__(self):
self.model = None
self.device = None
def load_model(self):
"""Loads the Whisper model, prioritizing GPU and handling memory."""
if self.model is None:
self.device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
logging.info(f"Using device: {self.device}")
model_size = "large-v2" if torch.cuda.is_available() else "medium"
logging.info(f"Loading Whisper model: {model_size}")
try:
self.model = whisper.load_model(model_size, device=self.device)
except RuntimeError as e:
if "out of memory" in str(e):
logging.error(f"Error: {e}")
logging.warning("Falling back to 'medium' model size due to memory constraints.")
self.model = whisper.load_model("medium", device=self.device)
else:
raise e
return self.model
def unload_model(self):
"""Unloads the model and clears CUDA cache."""
if self.model is not None:
del self.model
self.model = None
if torch.cuda.is_available():
torch.cuda.empty_cache()
logging.info("Model unloaded and CUDA cache cleared.")
async def transcribe_audio(audio_path, language, progress_bar,
task='transcribe', initial_prompt=None,
temperature=0.5, chunk_duration=30):
"""Transcribes audio using Whisper, handling chunking and errors."""
try:
model = WhisperModelCache.get_instance().load_model()
device = WhisperModelCache.get_instance().device
probe = ffmpeg.probe(audio_path)
total_duration = float(probe['format']['duration'])
num_chunks = int(total_duration // chunk_duration) + (total_duration % chunk_duration > 0)
progress_per_chunk = 20 / num_chunks
full_transcription = ""
for chunk_idx in range(num_chunks):
start_time = chunk_idx * chunk_duration
end_time = min((chunk_idx + 1) * chunk_duration, total_duration)
temp_chunk_path = f"{TEMP_FOLDER}/temp_chunk_{chunk_idx}.wav"
try:
(
ffmpeg
.input(audio_path)
.filter('atrim', start=start_time, end=end_time)
.output(temp_chunk_path, acodec='pcm_s16le', ac=1, ar='16k')
.overwrite_output()
.run(capture_stdout=True, capture_stderr=True)
)
except ffmpeg.Error as e:
logging.error(f"Error extracting audio chunk: {e.stderr.decode()}")
return "Error: Could not extract audio chunk for transcription"
result = await asyncio.to_thread(model.transcribe, temp_chunk_path,
language=language,
task=task,
initial_prompt=initial_prompt,
temperature=temperature)
full_transcription += result['text']
progress_bar.update(progress_per_chunk)
delete_temp_file(temp_chunk_path)
return full_transcription
except Exception as e:
logging.error(f"Error transcribing {audio_path}: {str(e)}")
return f"Error during transcription: {str(e)}"
# --- Anonymization ---
def anonymize_text(text):
"""Anonymizes personal information in text."""
text = re.sub(r'\b[A-Z][a-z]+ [A-Z][a-z]+\b|\S+@\S+|\d{3}[-.]?\d{3}[-.]?\d{4}',
lambda m: '[NAME]' if re.match(r'\b[A-Z][a-z]+ [A-Z][a-z]+\b', m.group()) else
'[EMAIL]' if '@' in m.group() else '[PHONE]',
text)
return text
# --- Gradio UI ---
async def process_audio(file, language, anonymize):
"""Processes audio: validation, conversion, transcription, anonymization, cleanup."""
try:
if file is None:
return "Error: Please upload an audio or video file."
if not is_supported_format(file):
raise ValueError(f"Unsupported file format: {file}")
progress_bar = tqdm(total=100, desc="Overall Process", unit="%", position=0, leave=True)
progress_bar.update(10)
temp_audio_path = convert_to_wav(file)
if not temp_audio_path:
raise ValueError(f"Failed to convert {file} to WAV format.")
progress_bar.update(30)
transcription = await transcribe_audio(temp_audio_path, language, progress_bar)
progress_bar.update(20)
delete_temp_file(temp_audio_path)
if anonymize:
transcription = anonymize_text(transcription)
progress_bar.update(10)
progress_bar.update(30)
progress_bar.close()
return transcription
except Exception as e:
print(f"Error processing audio: {e}")
return f"Error: {str(e)}"
def create_ui():
"""Create the Gradio UI."""
language_choices = ["en", "es", "fr", "de", "it", "pt", "nl", "ru", "zh", "ja", "ko", "ar", "he", "iw", "ar", "auto"]
output_format_choices = ["txt", "srt", "vtt", "tsv", "json"]
with gr.Blocks() as interface:
with gr.Row():
with gr.Column():
audio_input = gr.Audio(label="Upload Audio/Video", type="filepath")
task_dropdown = gr.Dropdown(
choices=["Transcribe", "Translate"],
label="Task",
value="Transcribe"
)
language_dropdown = gr.Dropdown(
choices=language_choices,
label="Language",
value="en", # Default to English
info="Select 'auto' for automatic language detection."
)
output_format_checkbox_group = gr.CheckboxGroup(
choices=output_format_choices,
label="Output Formats",
value=["txt"]
)
anonymize_checkbox = gr.Checkbox(label="Anonymize Transcription")
prompt_input = gr.Textbox(
label="Initial Prompt",
lines=2,
placeholder="Optional prompt to guide transcription"
)
temperature_slider = gr.Slider(
minimum=0.0,
maximum=1.0,
value=0.5,
label="Temperature"
)
timestamps_checkbox = gr.Checkbox(label="Include Word Timestamps")
transcribe_button = gr.Button(value="Transcribe")
with gr.Column():
transcription_output = gr.Textbox(label="Transcription", lines=10)
transcribe_button.click(
fn=process_audio,
inputs=[audio_input, language_dropdown, anonymize_checkbox],
outputs=transcription_output
)
return interface
# --- Main Execution ---
if __name__ == "__main__":
logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(levelname)s - %(message)s')
create_folders()
iface = create_ui()
iface.launch(debug=True, share=True) |