Delete whisper_server.py
Browse files- whisper_server.py +0 -63
whisper_server.py
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import os
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import tempfile
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from flask import request, jsonify
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from transformers import pipeline
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import torch
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import traceback
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# Define a writable directory for the model cache.
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# This now respects the HF_HOME environment variable set in the Dockerfile.
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cache_dir = os.environ.get("HF_HOME", "/tmp/.cache")
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os.makedirs(cache_dir, exist_ok=True)
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print("Loading openai/whisper-tiny model via transformers pipeline...")
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# Determine device
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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# Initialize the ASR pipeline with a more lightweight model
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# Using openai/whisper-tiny is much less memory intensive than the Hindi-specific one
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model = pipeline(
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"automatic-speech-recognition",
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model="openai/whisper-tiny",
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device=device,
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model_kwargs={"cache_dir": cache_dir}
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)
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print("Whisper model loaded.")
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def handle_transcribe():
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try:
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# Step 1: Validate request - looking for 'audio' key from frontend
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if 'audio' not in request.files:
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print("Error: 'audio' key not in request.files")
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return jsonify({'error': 'No audio file part in the request'}), 400
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file = request.files['audio']
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if file.filename == '':
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print("Error: No selected file")
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return jsonify({'error': 'No selected file'}), 400
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# Step 2: Use a temporary file to save the upload
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with tempfile.NamedTemporaryFile(delete=True, suffix=".webm") as temp_audio:
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file.save(temp_audio.name)
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print(f"Transcribing file: {temp_audio.name} with openai/whisper-tiny pipeline")
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# Step 3: Transcribe using the pipeline
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# The pipeline is robust and can handle various formats directly, leveraging ffmpeg.
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result = model(temp_audio.name)
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transcribed_text = result.get('text', '')
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print("Transcription successful.")
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return jsonify({'text': transcribed_text})
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except Exception as e:
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# Step 4: Robust error logging
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print("❌ Error in handle_transcribe():")
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traceback.print_exc()
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return jsonify({'error': f"An unexpected error occurred during transcription: {str(e)}"}), 500
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