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TalkEdit/utils/transcription.py

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import whisper
from pathlib import Path
from utils.audio_processing import extract_audio
import logging
import torch
import streamlit as st
try:
from utils.gpu_utils import configure_gpu, get_optimal_device
GPU_UTILS_AVAILABLE = True
except ImportError:
GPU_UTILS_AVAILABLE = False
try:
from utils.cache import load_from_cache, save_to_cache
CACHE_AVAILABLE = True
except ImportError:
CACHE_AVAILABLE = False
logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)
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WHISPER_MODEL = "base"
WHISPER_MODEL_SIZES = {
"tiny": 75,
"base": 140,
"small": 460,
"medium": 1500,
"large": 2900,
"large-v2": 2900,
"large-v3": 2900,
}
@st.cache_resource
def _load_whisper_model(model_name, device_str):
"""Load and cache a Whisper model. Cached across reruns."""
logger.info(f"Loading Whisper model: {model_name} on {device_str}")
device = torch.device(device_str)
try:
return whisper.load_model(model_name, device=device if device.type != "mps" else "cpu")
except (MemoryError, RuntimeError) as e:
err_str = str(e).lower()
if "out of memory" in err_str or "cannot allocate" in err_str or isinstance(e, MemoryError):
size_mb = WHISPER_MODEL_SIZES.get(model_name, "unknown")
raise MemoryError(
f"Not enough memory to load Whisper '{model_name}' model (~{size_mb}MB). "
f"Try a smaller model (tiny/base/small) or enable GPU acceleration."
) from e
raise
def transcribe_audio(audio_path: Path, model=WHISPER_MODEL, use_cache=True, cache_max_age=None,
use_gpu=True, memory_fraction=0.8):
"""
Transcribe audio using Whisper and return both segments and full transcript.
Args:
audio_path (Path): Path to the audio or video file
model (str): Whisper model size to use (tiny, base, small, medium, large)
use_cache (bool): Whether to use caching
cache_max_age (float, optional): Maximum age of cache in seconds
use_gpu (bool): Whether to use GPU acceleration if available
memory_fraction (float): Fraction of GPU memory to use (0.0 to 1.0)
Returns:
tuple: (segments, transcript) where segments is a list of dicts with timing info
"""
audio_path = Path(audio_path)
if use_cache and CACHE_AVAILABLE:
cached_data = load_from_cache(audio_path, model, "transcribe", cache_max_age)
if cached_data:
logger.info(f"Using cached transcription for {audio_path}")
return cached_data.get("segments", []), cached_data.get("transcript", "")
video_extensions = ['.mp4', '.avi', '.mov', '.mkv']
if audio_path.suffix.lower() in video_extensions:
audio_path = extract_audio(audio_path)
device = torch.device("cpu")
if use_gpu and GPU_UTILS_AVAILABLE:
gpu_config = configure_gpu(model, memory_fraction)
device = gpu_config["device"]
logger.info(f"Using device: {device} for transcription")
whisper_model = _load_whisper_model(model, str(device))
logger.info(f"Transcribing audio: {audio_path}")
result = whisper_model.transcribe(str(audio_path))
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transcript = result["text"]
segments = result["segments"]
if use_cache and CACHE_AVAILABLE:
cache_data = {
"transcript": transcript,
"segments": segments
}
save_to_cache(audio_path, cache_data, model, "transcribe")
return segments, transcript