""" Speech Recognition Module using Whisper Large-v3 Handles audio preprocessing and transcription """ import logging logger = logging.getLogger(__name__) import torch from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor from pydub import AudioSegment def transcribe_audio(audio_path): """ Convert audio file to text using Whisper ASR model Args: audio_path: Path to input audio file Returns: Transcribed English text """ logger.info(f"Starting transcription for: {audio_path}") try: # Audio conversion logger.debug("Converting audio format") audio = AudioSegment.from_file(audio_path) processed_audio = audio.set_frame_rate(16000).set_channels(1) wav_path = audio_path.replace(".mp3", ".wav") processed_audio.export(wav_path, format="wav") logger.debug(f"Audio converted to: {wav_path}") # Model initialization logger.info("Loading Whisper model") device = "cuda" if torch.cuda.is_available() else "cpu" logger.debug(f"Using device: {device}") model = AutoModelForSpeechSeq2Seq.from_pretrained( "openai/whisper-large-v3", torch_dtype=torch.float32, low_cpu_mem_usage=True, use_safetensors=True ).to(device) processor = AutoProcessor.from_pretrained("openai/whisper-large-v3") logger.debug("Model loaded successfully") # Processing logger.debug("Processing audio input") inputs = processor( wav_path, sampling_rate=16000, return_tensors="pt", truncation=True, chunk_length_s=30, stride_length_s=5 ).to(device) # Transcription logger.info("Generating transcription") with torch.no_grad(): outputs = model.generate(**inputs, language="en", task="transcribe") result = processor.batch_decode(outputs, skip_special_tokens=True)[0] logger.info(f"Transcription completed successfully") return result except Exception as e: logger.error(f"Transcription failed: {str(e)}", exc_info=True) raise