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Update model.py
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model.py
CHANGED
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import os
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import json
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import random
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import torch
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import
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from huggingface_hub import login, upload_folder
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from transformers.integrations import TensorBoardCallback
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from sklearn.metrics import accuracy_score, precision_recall_fscore_support
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from transformers import (
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Wav2Vec2FeatureExtractor,
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Trainer, TrainingArguments,
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EarlyStoppingCallback
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)
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MODEL = "ntu-spml/distilhubert" # modelo base utilizado, para usar otro basta con cambiar esto
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FEATURE_EXTRACTOR = Wav2Vec2FeatureExtractor.from_pretrained(MODEL)
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seed = 123
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MAX_DURATION = 1.00
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SAMPLING_RATE = FEATURE_EXTRACTOR.sampling_rate # 16000
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token = os.getenv(
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config_file = "models_config.json"
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clasificador = "class"
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monitor = "mon"
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batch_size = 16
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class AudioDataset(Dataset):
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def __init__(self, dataset_path, label2id):
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self.dataset_path = dataset_path
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self.label2id = label2id
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self.file_paths = []
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self.labels = []
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for label_dir, label_id in self.label2id.items():
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label_path = os.path.join(self.dataset_path, label_dir)
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if os.path.isdir(label_path):
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for file_name in os.listdir(label_path):
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audio_path = os.path.join(label_path, file_name)
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self.file_paths.append(audio_path)
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self.labels.append(label_id)
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def __len__(self):
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return len(self.file_paths)
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def __getitem__(self, idx):
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audio_path = self.file_paths[idx]
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label = self.labels[idx]
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input_values = self.preprocess_audio(audio_path)
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return {
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"input_values": input_values,
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"labels": torch.tensor(label)
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}
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def preprocess_audio(self, audio_path):
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waveform, sample_rate = torchaudio.load(
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audio_path,
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normalize=True, # Convierte a float32
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# num_frames= # TODO: Probar para que no haga falta recortar los audios
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)
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if sample_rate != SAMPLING_RATE: # Resamplear si no es 16kHz
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resampler = torchaudio.transforms.Resample(sample_rate, SAMPLING_RATE)
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waveform = resampler(waveform)
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if waveform.shape[0] > 1: # Si es stereo, convertir a mono
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waveform = waveform.mean(dim=0)
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waveform = waveform / (torch.max(torch.abs(waveform)) + 1e-6) # Sin 1e-6 el accuracy es pésimo!!
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max_length = int(SAMPLING_RATE * MAX_DURATION)
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if waveform.shape[0] > max_length:
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waveform = waveform[:max_length]
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else:
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# Pad the waveform if it's shorter than max length
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waveform = torch.nn.functional.pad(waveform, (0, max_length - waveform.shape[0]))
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inputs = FEATURE_EXTRACTOR(
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waveform,
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sampling_rate=SAMPLING_RATE,
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return_tensors="pt",
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# max_length=int(SAMPLING_RATE * MAX_DURATION),
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# truncation=True,
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padding=True,
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)
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return inputs.input_values.squeeze()
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def seed_everything():
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torch.manual_seed(seed)
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torch.cuda.manual_seed(seed)
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torch.backends.cudnn.deterministic = True
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torch.backends.cudnn.benchmark = False
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os.environ['CUBLAS_WORKSPACE_CONFIG'] = ':16384:8'
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def
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def
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dataset =
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train_dataset = torch.utils.data.Subset(dataset, train_indices)
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test_dataset = torch.utils.data.Subset(dataset, test_indices)
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train_dataloader = DataLoader(
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train_dataset, batch_size=batch_size, shuffle=shuffle, num_workers=num_workers, pin_memory=pin_memory
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)
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test_dataloader = DataLoader(
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test_dataset, batch_size=batch_size, shuffle=shuffle, num_workers=num_workers, pin_memory=pin_memory
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)
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return train_dataloader, test_dataloader, label2id, id2label
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def load_model(num_labels, label2id, id2label):
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MODEL,
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num_labels=num_labels,
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label2id=label2id,
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id2label=id2label
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finetuning_task="audio-classification"
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)
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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model = HubertForSequenceClassification.from_pretrained( # TODO: mirar parámetros. Posibles optimizaciones
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MODEL,
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config=config,
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torch_dtype=torch.float32, # No afecta 1ª época, mejor ponerlo
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)
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model.to(device)
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return model
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def model_params(dataset_path):
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def compute_metrics(eval_pred):
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predictions =
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references =
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accuracy = accuracy_score(references, predictions)
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precision, recall, f1, _ = precision_recall_fscore_support(references, predictions, average='weighted')
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return {
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"accuracy":
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"precision": precision,
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"recall": recall,
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"f1": f1,
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}
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def
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trainer = Trainer(
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model=model,
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args=training_args,
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compute_metrics=compute_metrics,
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train_dataset=
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eval_dataset=
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callbacks=[
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)
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torch.cuda.empty_cache() # liberar memoria de la GPU
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trainer.train() # se pueden modificar los parámetros para continuar el train
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login(token, add_to_git_credential=True)
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trainer.push_to_hub(token=token) # Subir modelo a mi cuenta. Necesario para hacer la predicción, no sé por qué.
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trainer.save_model(output_dir) # para subir el modelo a Hugging Face. Necesario para hacer la predicción, no sé por qué.
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os.makedirs(output_dir, exist_ok=True) # Crear carpeta con el modelo si no existe
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def load_config(model_name):
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with open(config_file, 'r') as f:
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return model_config
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if __name__ == "__main__":
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config = load_config(monitor) # PARA
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training_args = config["training_args"]
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output_dir = config["output_dir"]
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dataset_path = config["dataset_path"]
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import torch
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import numpy as np
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import os
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from huggingface_hub import login, upload_folder
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from datasets import load_dataset, Audio
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from transformers.integrations import TensorBoardCallback
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from transformers import (
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Wav2Vec2FeatureExtractor, AutoModelForAudioClassification,
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Trainer, TrainingArguments,
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EarlyStoppingCallback
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)
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import json
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# SE USA FLOAT32 EN EL MODELO ORIGINAL
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MODEL = "ntu-spml/distilhubert" # modelo base utilizado, para usar otro basta con cambiar esto
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FEATURE_EXTRACTOR = Wav2Vec2FeatureExtractor.from_pretrained(MODEL)
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seed = 123
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MAX_DURATION = 1.00
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SAMPLING_RATE = FEATURE_EXTRACTOR.sampling_rate # 16000 # antes estaba float16
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token = os.getenv('MODEL_REPO_ID')
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config_file = "models_config.json"
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clasificador = "class"
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monitor = "mon"
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def seed_everything():
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np.random.seed(seed)
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torch.manual_seed(seed)
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torch.cuda.manual_seed(seed)
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torch.backends.cudnn.deterministic = True
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torch.backends.cudnn.benchmark = False
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os.environ['CUBLAS_WORKSPACE_CONFIG'] = ':16384:8'
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def preprocess_audio(audio_arrays, batch=True):
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if batch:
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audios = [x["array"] for x in audio_arrays["audio"]] # para usar aquí
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else:
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audios = [audio_arrays] # para usar en realtime.py
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inputs = Wav2Vec2FeatureExtractor.from_pretrained(MODEL)(
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raw_speech=audios,
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sampling_rate=SAMPLING_RATE,
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return_tensors="pt", # Devolver tensores de PyTorch
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max_length=int(SAMPLING_RATE * MAX_DURATION), # Necesario para truncation
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truncation=True, # Muchísimo más rápido.
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padding=True, # Vectores igual longitud
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do_normalize=True, # No afecta 1ª época, no sé si necesario
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# return_attention_mask=True, # Empeora 1ª época. No sé si necesario
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padding_value=0.0, # No afecta 1ª época, no sé si necesario
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float=32 # No afecta 1ª época, no sé si necesario
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)
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return inputs
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def load_and_prepare_dataset(dataset_path):
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dataset = load_dataset(dataset_path, split="train") # Split para que no ponga train de primeras
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# dataset = dataset.cast_column("audio", Audio(sampling_rate=SAMPLING_RATE)) # Da mejor accuracy pero creo que cambia el preprocesado.
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encoded_dataset = dataset.map(preprocess_audio, remove_columns=["audio"], batched=True) # num_proc hace q no vaya realtime
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labels = encoded_dataset.features["label"].names
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label2id = {label: str(i) for i, label in enumerate(labels)}
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id2label = {str(i): label for i, label in enumerate(labels)}
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encoded_dataset = encoded_dataset.train_test_split(test_size=0.2, seed=seed, stratify_by_column="label")
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return encoded_dataset, label2id, id2label
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def load_model(num_labels, label2id, id2label):
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model = AutoModelForAudioClassification.from_pretrained(
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MODEL,
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num_labels=num_labels,
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label2id=label2id,
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id2label=id2label
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)
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return model
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def model_params(dataset_path):
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login(token, add_to_git_credential=True)
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seed_everything()
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encoded_dataset, label2id, id2label = load_and_prepare_dataset(dataset_path)
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model = load_model(len(id2label), label2id, id2label)
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return model, encoded_dataset, id2label
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def compute_metrics(eval_pred):
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predictions = np.argmax(eval_pred.predictions, axis=1)
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references = eval_pred.label_ids
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return {
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"accuracy": np.mean(predictions == references),
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}
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def model_training(training_args, output_dir, dataset_path):
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model, encoded_dataset, _ = model_params(dataset_path)
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tensorboard_callback = TensorBoardCallback()
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early_stopping_callback = EarlyStoppingCallback(early_stopping_patience=3)
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trainer = Trainer(
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model=model,
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args=training_args,
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compute_metrics=compute_metrics,
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train_dataset=encoded_dataset["train"],
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eval_dataset=encoded_dataset["test"],
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callbacks=[tensorboard_callback, early_stopping_callback]
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)
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torch.cuda.empty_cache() # liberar memoria de la GPU
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trainer.train() # se pueden modificar los parámetros para continuar el train
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trainer.push_to_hub(token=token) # Subir modelo a mi cuenta. Necesario para hacer la predicción, no sé por qué.
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trainer.save_model(output_dir) # para subir el modelo a Hugging Face. Necesario para hacer la predicción, no sé por qué.
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os.makedirs(output_dir, exist_ok=True) # Crear carpeta con el modelo si no existe
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upload_folder(repo_id=f"A-POR-LOS-8000/{output_dir}",folder_path=output_dir, token=token) # subir modelo a organización
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def load_config(model_name):
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with open(config_file, 'r') as f:
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return model_config
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if __name__ == "__main__":
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config = load_config(clasificador) # PARA CAMBIAR MODELOS
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# config = load_config(monitor) # PARA CAMBI
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training_args = config["training_args"]
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output_dir = config["output_dir"]
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dataset_path = config["dataset_path"]
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model_training(training_args, output_dir, dataset_path)
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