Added transcription (needs to be refactored)
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# Python-generated files
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__pycache__/
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*.py[oc]
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build/
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dist/
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wheels/
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*.egg-info
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# Virtual environments
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.venv
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main.todo
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sample.mp3
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@@ -0,0 +1,126 @@
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from transformers import WhisperProcessor, WhisperForConditionalGeneration
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import torch
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import torchaudio
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import logging
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import time
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import asyncio
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FILENAME = "sample"
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SAMPLING_FREQUENCY = 16000
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logging.basicConfig(
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level=logging.INFO,
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format='%(asctime)s - %(name)s - %(levelname)s - %(message)s'
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)
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logger = logging.getLogger(__name__)
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print("=== Checking PyTorch ===")
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print(f"Torch version: {torch.version}")
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print(f"CUDA is available: {torch.cuda.is_available()}")
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if torch.cuda.is_available():
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print(f"CUDA version: {torch.version.cuda}")
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print(f"Number of GPU: {torch.cuda.device_count()}")
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print(f"Name of GPU: {torch.cuda.get_device_name(0)}")
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print("=== Check completed ===")
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device = "cuda"
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logger.info("Loading model WhisperProcessor...")
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processor = WhisperProcessor.from_pretrained("openai/whisper-large-v2")
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logger.info("Loading model WhisperForConditionalGeneration...")
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model = WhisperForConditionalGeneration.from_pretrained(
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"openai/whisper-large-v2",
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torch_dtype=torch.float16,
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device_map="auto"
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).to(device)
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logger.info("Model loaded")
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def transcribe_long(path: str, language="ru", chunk_length_s: int = 30):
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logger.info(f"Starting transcription of long file: {path}")
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start_time = time.time()
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try:
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waveform, sr = torchaudio.load(path, format="mp3", backend="ffmpeg")
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if waveform.shape[0] > 1:
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waveform = waveform.mean(dim=0, keepdim=True)
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waveform = torchaudio.functional.resample(waveform, sr, SAMPLING_FREQUENCY).squeeze()
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total_samples = waveform.shape[0]
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chunk_size = chunk_length_s * SAMPLING_FREQUENCY
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num_chunks = (total_samples + chunk_size - 1) // chunk_size
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logger.info(f"File length - {total_samples/SAMPLING_FREQUENCY:.1f} seconds, splitting on {num_chunks} chunks by {chunk_length_s} seconds")
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transcripts = []
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for i in range(num_chunks):
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start = i * chunk_size
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end = min((i + 1) * chunk_size, total_samples)
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chunk = waveform[start:end].cpu().numpy()
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inputs = processor(chunk, sampling_rate=SAMPLING_FREQUENCY, return_tensors="pt")
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input_features = inputs.input_features.to(device).to(torch.float16)
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with torch.no_grad():
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predicted_ids = model.generate(
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input_features,
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language=language,
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task="transcribe",
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temperature=0.0
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)
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text = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
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transcripts.append(text)
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logger.info(f"Чанк {i+1}/{num_chunks} готов ({(end/SAMPLING_FREQUENCY):.1f} сек)")
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end_time = time.time()
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logger.info(f"Transcription completed - {end_time - start_time:.2f} seconds")
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return " ".join(transcripts)
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except Exception as e:
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logger.error(f"Transcription error: {str(e)}")
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raise
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def split_into_chunks(
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filepath: str,
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chunk_length_s: int = 30
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) -> list:
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try:
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chunks = []
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waveform, sr = torchaudio.load(filepath, format="mp3", backend="ffmpeg")
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if waveform.shape[0] > 1:
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waveform = waveform.mean(dim=0, keepdim=True)
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logger.info(f"Started splitting file into chunks with length {chunk_length_s}")
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total_samples = waveform.shape[0]
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chunk_size = chunk_length_s
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num_chunks = (total_samples + chunk_size - 1) // chunk_size
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logger.info(f"File length - {total_samples/SAMPLING_FREQUENCY:.1f} seconds, разобьём на {num_chunks} чанков по {chunk_length_s} секунд")
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for i in range(num_chunks):
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start = i * chunk_size
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end = min((i + 1) * chunk_size, total_samples)
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chunk = waveform[start:end].cpu().numpy()
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chunks.append(chunk)
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return chunks
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except Exception as e:
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logger.error(f"Error while splitting to chunks: {str(e)}")
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raise
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chunks = split_into_chunks(f"{FILENAME}.mp3")
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try:
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result = transcribe_long(f"{FILENAME}.mp3")
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print(result)
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except Exception as e:
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logger.error(f"Execution error: {e}")
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@@ -0,0 +1,8 @@
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# python -m venv .venv
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# pip install -r requirements.txt --extra-index-url https://download.pytorch.org/whl/cu128
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conda init -c speech-to-conspect
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conda install pytorch torchvision torchaudio -c pytorch -c nvidia
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conda install accelerate transformers ffmpeg -c conda-forge
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