update audio classification script

audio_classification_scripts/run_mms_lid.sh

 #!/usr/bin/env bash
 
-python run_audio_classification.py \
-    --model_name_or_path "facebook/mms-lid-126" \
+CUDA_VISIBLE_DEVICES=2 python run_audio_classification_one_layer.py \
+    --model_name_or_path "facebook/mms-lid-4017" \
     --train_dataset_name "stable-speech/concatenated-normalized-accent-dataset" \
     --train_dataset_config_name "default" \
     --train_split_name "train" \
@@ -10,11 +10,11 @@ python run_audio_classification.py \
     --eval_dataset_config_name "default" \
     --eval_split_name "test" \
     --eval_label_column_name "labels" \
-    --output_dir "./" \
+    --output_dir "./tmp/" \
     --do_train \
     --do_eval \
     --overwrite_output_dir \
-    --remove_unused_columns False \
+    --remove_unused_columns false \
     --fp16 \
     --fp16_full_eval \
     --learning_rate 1e-4 \
@@ -30,9 +30,11 @@ python run_audio_classification.py \
     --logging_strategy "steps" \
     --logging_steps 10 \
     --evaluation_strategy "steps" \
-    --eval_steps 500 \
+    --eval_steps 300 \
     --save_strategy "no" \
     --save_steps 2000 \
-    --freeze_base_model True \
-    --push_to_hub False \
-    --trust_remote_code
+    --freeze_base_model true \
+    --freeze_feature_encoder true \
+    --push_to_hub false \
+    --trust_remote_code \
+    --use_weighted_layer_sum true \

audio_classification_scripts/run_sweep.yaml

 command:
   - python3
   - ${program}
-  - --load_best_model_at_end
   - --fp16
   - --fp16_full_eval
   - --do_train
   - --do_eval
   - --trust_remote_code
   - --overwrite_output_dir
-  - --ignore_mismatched_sizes
-  - --gradient_checkpointing
   - ${args}
-method: grid
+method: random
 metric:
   goal: maximize
   name: eval/accuracy
 parameters:
   model_name_or_path:
-    value: facebook/mms-lid-126
+    value: facebook/mms-lid-4017
   train_dataset_name:
-    value: stable-speech/concatenated-accent-dataset
+    value:  "stable-speech/concatenated-normalized-accent-dataset+stable-speech/concatenated-common-voice-15-accented"
   train_dataset_config_name:
-    value: default
+    value: "default+default"
   train_split_name:
-    value: train
+    value: "train+train"
   train_label_column_name:
-    value: labels
+    value: "labels+labels"
   eval_dataset_name:
-    value: stable-speech/concatenated-accent-dataset
+    value: stable-speech/concatenated-normalized-accent-dataset
   eval_dataset_config_name:
     value: default
   eval_split_name:
@@ -35,31 +32,38 @@ parameters:
   eval_label_column_name:
     value: labels
   output_dir:
-    value: ./
+    value: "/raid/yoach/tmp/"
   remove_unused_columns:
     value: false
   learning_rate:
-    value: 1e-4
+    distribution: log_uniform_values
+    min: 3e-6
+    max: 0.01
   lr_scheduler_type:
-    value: constant_with_warmup
+    value: constant
   max_length_seconds:
     value: 20  # give some data diversity for longer audio samples
   min_length_seconds:
-    value: 7
+    value: 5
   attention_mask:
-    value: true
-  warmup_steps:
-    value: 100
-  max_steps:
-    value: 2000
+    values:
+    - true
+  num_train_epochs:
+    values:
+      - 2
+      - 5
+      - 10
+      - 20
+      - 40
+      - 60
   per_device_train_batch_size:
     value: 32
   per_device_eval_batch_size:
-    value: 16
+    value: 32
   preprocessing_num_workers:
-    value: 4
+    value: 8
   dataloader_num_workers:
-    value: 4
+    value: 8
   logging_strategy:
     value: steps
   logging_steps:
@@ -67,20 +71,28 @@ parameters:
   evaluation_strategy:
     value: steps
   eval_steps:
-    value: 1000
+    value: 2000
   save_strategy:
-    value: steps
+    value: "no"
   save_steps:
     value: 2000
   metric_for_best_model:
     value: accuracy
-  freeze_base_model:
-    values:
-     - false
-     - true
-  group_by_length:
-    value: false  # TODO(SG): batch by length
   push_to_hub:
     value: false
+  use_weighted_layer_sum:
+    value: false
+  freeze_base_model:
+    value: true
+  max_samples_per_label:
+    value: 10000
+  save_to_disk:
+    value: "/raid/yoach/tmp_dataset_accents/" 
+  temporary_save_to_disk:
+    value: "/raid/yoach/tmp_hidden_states/"
+  use_last_embedding_layer:
+    value: true
+  filter_threshold:
+    value: "0.001"
 program: run_audio_classification.py
-project: mms-lid-accent-classification
+project: mms-lid-accent-classification-v2
\ No newline at end of file

run_audio_classification.py

@@ -21,7 +21,9 @@ import sys
 from collections import Counter
 from dataclasses import dataclass, field
 from random import randint
-from typing import List, Optional, Union
+from typing import List, Optional, Union, Dict
+
+import torch
 
 import datasets
 import evaluate
@@ -41,7 +43,14 @@ from transformers import (
 from transformers.models.whisper.tokenization_whisper import LANGUAGES
 from transformers.trainer_utils import get_last_checkpoint
 from transformers.utils import check_min_version
+from transformers.trainer_pt_utils import LengthGroupedSampler
 
+from transformers import Wav2Vec2ForSequenceClassification, Wav2Vec2Model
+from transformers import Wav2Vec2BertForSequenceClassification, Wav2Vec2BertModel
+from transformers.models.wav2vec2.modeling_wav2vec2 import _HIDDEN_STATES_START_POSITION
+from transformers.modeling_outputs import SequenceClassifierOutput
+from torch import nn
+from torch.nn import CrossEntropyLoss
 
 logger = logging.getLogger(__name__)
 
@@ -65,6 +74,81 @@ def deterministic_subsample(wav: np.ndarray, max_length: float, sample_rate: int
         return wav
     return wav[0:sample_length]
 
+class SequenceClassificationModel(Wav2Vec2ForSequenceClassification):
+    def __init__(self, config):
+        super().__init__(config)
+
+        if hasattr(config, "add_adapter") and config.add_adapter:
+            raise ValueError(
+                "Sequence classification does not support the use of Wav2Vec2 adapters (config.add_adapter=True)"
+            )
+        self.wav2vec2 = Wav2Vec2Model(config)
+        num_layers = config.num_hidden_layers + 1  # transformer layers + input embeddings
+        if config.use_weighted_layer_sum:
+            self.layer_weights = nn.Parameter(torch.ones(num_layers) / num_layers)
+        self.classifier = nn.Linear(config.hidden_size, config.num_labels)
+        
+        # To bypass w2v2
+        self.compute_w2v2 = True
+
+        # Initialize weights and apply final processing
+        self.post_init()
+        
+    def forward(
+        self,
+        input_features=None,
+        attention_mask=None,
+        output_attentions=None,
+        output_hidden_states=None,
+        return_dict=None,
+        labels=None,
+        hidden_states=None, # added
+    ):
+        r"""
+        labels (`torch.LongTensor` of shape `(batch_size,)`, *optional*):
+            Labels for computing the sequence classification/regression loss. Indices should be in `[0, ...,
+            config.num_labels - 1]`. If `config.num_labels == 1` a regression loss is computed (Mean-Square loss), If
+            `config.num_labels > 1` a classification loss is computed (Cross-Entropy).
+        """
+
+        return_dict = return_dict if return_dict is not None else self.config.use_return_dict
+        output_hidden_states = True
+
+        if self.compute_w2v2:
+            outputs = self.wav2vec2(
+                input_features,
+                attention_mask=attention_mask,
+                output_attentions=output_attentions,
+                output_hidden_states=output_hidden_states,
+                return_dict=return_dict,
+            )
+
+            hidden_states = outputs[_HIDDEN_STATES_START_POSITION][-1] # take last embedding layer
+
+            if attention_mask is None:
+                pooled_output = hidden_states.mean(dim=1)
+            else:
+                padding_mask = self._get_feature_vector_attention_mask(hidden_states.shape[1], attention_mask)
+                hidden_states[~padding_mask] = 0.0
+                pooled_output = hidden_states.sum(dim=1) / padding_mask.sum(dim=1).view(-1, 1)
+        else:
+            pooled_output = hidden_states
+
+        logits = self.classifier(pooled_output)
+
+        loss = None
+        if labels is not None:
+            loss_fct = CrossEntropyLoss()
+            loss = loss_fct(logits.view(-1, self.config.num_labels), labels.view(-1))
+
+        if not return_dict:
+            output = (logits,) + outputs[_HIDDEN_STATES_START_POSITION:]
+            return ((loss,) + output) if loss is not None else output
+
+        return SequenceClassifierOutput(
+            loss=loss,
+            logits=logits,
+        )
 
 # This list first defines the accent prefixes, which we use to strip the accent from CV
 # e.g. England, southern accent, slight west-country expression -> England
@@ -84,7 +168,8 @@ STARTS_WITH = [
     "German",
     "Filipino",
     "India",
-    "Irish" "Israeli",
+    "Irish",
+    "Israeli",
     "Italian",
     "Japanese",
     "Kenyan",
@@ -253,6 +338,41 @@ def preprocess_labels(label: str) -> str:
     return label
 
     
+@dataclass
+class DataCollatorFeatureExtractorWithPadding:
+    """
+    Data collator that will dynamically pad the inputs received to the longest sequence in the batch.
+    """
+
+    feature_extractor: AutoFeatureExtractor
+    max_length_seconds: int
+    feature_extractor_input_name: Optional[str] = "input_values"
+
+    def __call__(self, features: List[Dict[str, Union[List[int], torch.Tensor]]]) -> Dict[str, torch.Tensor]:
+        # split inputs and labels since they have to be of different lengths and need
+        # different padding methods
+        audios = [torch.tensor(deterministic_subsample(feature["audio"]["array"], max_length=self.max_length_seconds, sample_rate=self.feature_extractor.sampling_rate)).squeeze().numpy() for feature in features]
+        batch = self.feature_extractor(audios, return_tensors="pt", padding="longest", return_attention_mask=True)
+        
+        batch["labels"] = torch.tensor([feature["labels_id"] for feature in features])
+        return batch
+
+@dataclass
+class DataCollatorHiddenStatesPadding:
+    """
+    Data collator that will dynamically pad the inputs received to the longest sequence in the batch.
+    """
+
+    def __call__(self, features: List[Dict[str, Union[List[int], torch.Tensor]]]) -> Dict[str, torch.Tensor]:        
+        hidden_states = torch.stack([torch.tensor(feature["hidden_states"]) for feature in features])
+
+        batch = {"hidden_states": hidden_states}
+        
+        batch["labels"] = torch.tensor([feature["labels_id"] for feature in features])
+        return batch
+
+
+
 @dataclass
 class DataTrainingArguments:
     """
@@ -359,6 +479,26 @@ class DataTrainingArguments:
         default=1.0,
         metadata={"help": "Filter labels that occur less than `filter_threshold` percent in the training/eval data."},
     )
+    max_samples_per_label: Optional[int] = field(
+        default=None,
+        metadata={
+            "help": (
+                "If set, randomly limits the number of samples per label."
+            )
+        },   
+    )
+    save_to_disk: str = field(
+        default=None,
+        metadata={
+            "help": "If set, will save the dataset to this path if this is an empyt folder. If not empty, will load the datasets from it."
+        }
+    )
+    temporary_save_to_disk: str = field(
+        default=None,
+        metadata={
+            "help": "Temporarily save audio labels here."
+        }
+    )
 
 
 @dataclass
@@ -369,7 +509,7 @@ class ModelArguments:
 
     model_name_or_path: str = field(
         default="facebook/wav2vec2-base",
-        metadata={"help": "Path to pretrained model or model identifier from huggingface.co/models"},
+        metadata={"help": "Path to pretrained model or model identifier from huggingface.co/models. Only works with Wav2Vec2 models"},
     )
     config_name: Optional[str] = field(
         default=None, metadata={"help": "Pretrained config name or path if not the same as model_name"}
@@ -465,6 +605,8 @@ class ModelArguments:
         metadata={"help": "Length of vector span to mask along the feature axis."},
     )
     layerdrop: float = field(default=0.0, metadata={"help": "The LayerDrop probability."})
+    use_weighted_layer_sum: bool = field(default=False, metadata={"help": "Whether to use a weighted average of layer outputs with learned weights."})
+    use_last_embedding_layer: bool = field(default=False, metadata={"help": "Whether to use the last layer hidden state. Only work with W2V model."})
 
 
 def convert_dataset_str_to_list(
@@ -661,10 +803,16 @@ def main():
                 "the `--output_dir` or add `--overwrite_output_dir` to train from scratch."
             )
 
+    if data_args.save_to_disk is not None:
+        os.makedirs(data_args.save_to_disk, exist_ok=True)
+    
+    # assume that the dataset has been saved to `save_to_disk` if the latter is not empty
+    dataset_was_precomputed = len(os.listdir(data_args.save_to_disk)) > 0
+    if dataset_was_precomputed:
+        raw_datasets = datasets.load_from_disk(data_args.save_to_disk)
+    else:
         # Initialize our dataset and prepare it for the audio classification task.
         raw_datasets = DatasetDict()
-    # set seed for determinism
-    set_seed(training_args.seed)
 
         if training_args.do_train:
             raw_datasets["train"] = load_multiple_datasets(
@@ -742,12 +890,16 @@ def main():
         trust_remote_code=model_args.trust_remote_code,
     )
     
+    feature_extractor_input_name = feature_extractor.model_input_names[0]
+
+    if not dataset_was_precomputed:
         # `datasets` takes care of automatically loading and resampling the audio,
         # so we just need to set the correct target sampling rate.
         raw_datasets = raw_datasets.cast_column(
             data_args.audio_column_name, datasets.features.Audio(sampling_rate=feature_extractor.sampling_rate)
         )
 
+        with training_args.main_process_first():
             if training_args.do_train:
                 if data_args.max_train_samples is not None:
                     raw_datasets["train"] = (
@@ -763,23 +915,33 @@ def main():
     sampling_rate = feature_extractor.sampling_rate
     model_input_name = feature_extractor.model_input_names[0]
 
-    def prepare_dataset(batch):
-        batch["length"] = len(batch["audio"]["array"])
-        batch["labels"] = preprocess_labels(batch["labels"])
+    if not dataset_was_precomputed:
+        def prepare_dataset(audio, labels):
+            batch = {}
+            batch["length"] = len(audio["array"])
+            batch["labels"] = preprocess_labels(labels)
             return batch
 
-    raw_datasets = raw_datasets.map(
+        with training_args.main_process_first():
+            tmp_datasets = raw_datasets.map(
                 prepare_dataset,
                 num_proc=data_args.preprocessing_num_workers,
+                input_columns=["audio", "labels"],
+                remove_columns=[col for col  in next(iter(raw_datasets.values())).column_names if col != "labels"], # this is a trick to avoid to rewrite the entire audio column which takes ages
                 desc="Computing audio length",
             )
             
+        for split in raw_datasets:
+            raw_datasets[split] = concatenate_datasets([raw_datasets[split].remove_columns(["labels"]), tmp_datasets[split]], axis=1)
+
     # filter training data with inputs < min_input_length
     min_input_length = data_args.min_length_seconds * sampling_rate
 
+    if not dataset_was_precomputed:
         def is_audio_valid(input_length):
             return input_length > min_input_length
 
+        with training_args.main_process_first():
             raw_datasets = raw_datasets.filter(
                 is_audio_valid,
                 input_columns=["length"],
@@ -791,6 +953,7 @@ def main():
         def is_label_valid(label):
             return label != "Unknown"
 
+        with training_args.main_process_first():
             raw_datasets = raw_datasets.filter(
                 is_label_valid,
                 input_columns=["labels"],
@@ -798,6 +961,20 @@ def main():
                 desc="Filtering by labels",
             )
         
+        if training_args.do_train and data_args.max_samples_per_label:
+            label_names = set(raw_datasets["train"]["labels"])
+            labels = np.array(raw_datasets["train"]["labels"])
+            indices = np.arange(len(labels))
+            indices_to_keep = []
+            set_seed(training_args.seed)
+            for label in label_names:
+                label_indices = indices[labels==label]
+                label_indices = np.random.choice(label_indices, size=min(data_args.max_samples_per_label, len(label_indices)), replace=False)
+                indices_to_keep.extend(np.random.choice(label_indices, size=min(data_args.max_samples_per_label, len(label_indices)), replace=False))
+                    
+            with training_args.main_process_first():
+                raw_datasets["train"] = raw_datasets["train"].select(indices_to_keep).shuffle(seed=training_args.seed)
+
         # Print a summary of the labels to the stddout (helps identify low-label classes that could be filtered)
         # sort by freq
         count_labels_dict = Counter(raw_datasets["train"]["labels"])
@@ -819,6 +996,9 @@ def main():
             def is_label_valid(label):
                 return label not in labels_to_remove
 
+            logger.info(f"Labels removed: {labels_to_remove}")
+            
+            with training_args.main_process_first():
                 raw_datasets = raw_datasets.filter(
                     is_label_valid,
                     input_columns=["labels"],
@@ -829,37 +1009,42 @@ def main():
         # We'll include these in the model's config to get human readable labels in the Inference API.
         set_labels = set(raw_datasets["train"]["labels"])
         if training_args.do_eval:
+            logger.info(f"The following accents are present in the eval set but not the test set: {set(raw_datasets['eval']['labels']) - set_labels}")
             set_labels = set_labels.union(set(raw_datasets["eval"]["labels"]))
         label2id, id2label = {}, {}
-    for i, label in enumerate(set(set_labels)):
+        for i, label in enumerate(sorted(list(set_labels))):
             label2id[label] = str(i)
             id2label[str(i)] = label
             
-    def train_transforms(batch):
-        """Apply train_transforms across a batch."""
-        subsampled_wavs = []
-        for audio in batch["audio"]:
-            wav = deterministic_subsample(
-                audio["array"], max_length=data_args.max_length_seconds, sample_rate=feature_extractor.sampling_rate
-            )
-            subsampled_wavs.append(wav)
-        inputs = feature_extractor(
-            subsampled_wavs, return_attention_mask=model_args.attention_mask, sampling_rate=sampling_rate
-        )
-        output_batch = {
-            model_input_name: inputs.get(model_input_name),
-            "attention_mask": inputs.get("attention_mask"),
-            "labels": [int(label2id[label]) for label in batch["labels"]],
-        }
-        return output_batch
+        with training_args.main_process_first():
+            raw_datasets = raw_datasets.map(
+                    lambda label: {"labels_id": int(label2id[label])},
+                    input_columns=["labels"],
+                    num_proc=data_args.preprocessing_num_workers,
+                    desc="Apply labels id",
+                )
+    else:
+        label2id, id2label = {}, {}
 
+        # TODO: slow - probably not best
         if training_args.do_train:
-        # Set the training transforms
-        raw_datasets["train"].set_transform(train_transforms, output_all_columns=False)
-
+            for sample in raw_datasets["train"]:
+                candidate = label2id.get(sample["labels"])
+                if candidate is not None and candidate != sample["labels_id"]:
+                    print(f"issue {candidate} should be {sample['labels_id']}")
+                label2id[sample["labels"]] = sample["labels_id"]
         if training_args.do_eval:
-        # Set the validation transforms
-        raw_datasets["eval"].set_transform(train_transforms, output_all_columns=False)
+            for sample in raw_datasets["eval"]:
+                candidate = label2id.get(sample["labels"])
+                if candidate is not None and candidate != sample["labels_id"]:
+                    print(f"issue {candidate} should be {sample['labels_id']}")
+                label2id[sample["labels"]] = sample["labels_id"]
+                
+        # if training_args.do_train:
+        #     label2id = dict(zip(raw_datasets["train"]["labels"], raw_datasets["train"]["labels_id"]))
+        # if training_args.do_eval:
+        #     label2id.update(dict(zip(raw_datasets["eval"]["labels"], raw_datasets["eval"]["labels_id"])))
+        id2label = {str(val): key for (key, val) in label2id.items()}
         
     # Load the accuracy metric from the datasets package
     metric = evaluate.load("accuracy", cache_dir=model_args.cache_dir)
@@ -895,10 +1080,12 @@ def main():
             "mask_feature_length": model_args.mask_feature_length,
             "layerdrop": model_args.layerdrop,
             "activation_dropout": model_args.activation_dropout,
+            "use_weighted_layer_sum": model_args.use_weighted_layer_sum,
         }
     )
 
-    model = AutoModelForAudioClassification.from_pretrained(
+    if model_args.use_last_embedding_layer:
+        model = SequenceClassificationModel.from_pretrained(
             model_args.model_name_or_path,
             from_tf=bool(".ckpt" in model_args.model_name_or_path),
             config=config,
@@ -909,6 +1096,19 @@ def main():
             ignore_mismatched_sizes=model_args.ignore_mismatched_sizes,
         )
         
+        if model_args.freeze_base_model: 
+            model.compute_w2v2 = False
+    else:
+        model = AutoModelForAudioClassification.from_pretrained(
+            model_args.model_name_or_path,
+            from_tf=bool(".ckpt" in model_args.model_name_or_path),
+            config=config,
+            cache_dir=model_args.cache_dir,
+            revision=model_args.model_revision,
+            token=model_args.token,
+            trust_remote_code=model_args.trust_remote_code,
+            ignore_mismatched_sizes=model_args.ignore_mismatched_sizes,
+        )
     # freeze the convolutional waveform encoder for wav2vec2-style models
     if model_args.freeze_feature_encoder:
         if hasattr(model, "freeze_feature_encoder"):
@@ -928,15 +1128,91 @@ def main():
         else:
             raise ValueError("Method for freezing the base module of the audio encoder is not defined")
 
+    if model_args.freeze_base_model and model_args.use_last_embedding_layer:
+        if not dataset_was_precomputed:
+            # precomputing hidden states
+            from torch.utils.data import DataLoader
+            from accelerate import Accelerator
+            
+            _HIDDEN_STATES_START_POSITION = 2
+            
+            if training_args.fp16:
+                mixed_precision = "fp16"
+            elif training_args.bf16:
+                mixed_precision = "bf16"
+            else:
+                mixed_precision = "no"
+
+            accelerator = Accelerator(
+                gradient_accumulation_steps=training_args.gradient_accumulation_steps,
+                mixed_precision=mixed_precision,
+                project_dir=training_args.output_dir,
+            )
+            
+            audio_data_collator = DataCollatorFeatureExtractorWithPadding(feature_extractor, max_length_seconds=data_args.max_length_seconds, feature_extractor_input_name=feature_extractor_input_name)
+
+            for split in raw_datasets:
+                data_loader = DataLoader(
+                    raw_datasets[split],
+                    batch_size=training_args.per_device_train_batch_size, # TODO: chose another one
+                    collate_fn=audio_data_collator,
+                    num_workers=training_args.dataloader_num_workers,
+                    pin_memory=True,
+                )
+                data_loader = accelerator.prepare(data_loader)        
+                
+                all_encoder_outputs = []
+                for batch in tqdm(data_loader, disable=not accelerator.is_local_main_process):
+                    model.wav2vec2.to(batch[feature_extractor_input_name].device)
+                    
+                    with torch.no_grad():
+                        encoder_outputs = model.wav2vec2(batch[feature_extractor_input_name], attention_mask=batch.get("attention_mask", None), output_hidden_states=True)
+                    hidden_states = encoder_outputs[_HIDDEN_STATES_START_POSITION][-1]
+                    if batch.get("attention_mask", None) is None:
+                        hidden_states = hidden_states.mean(dim=1)
+                    else:
+                        padding_mask = model._get_feature_vector_attention_mask(hidden_states.shape[1], batch.get("attention_mask", None))
+                        hidden_states[~padding_mask] = 0.0
+                        hidden_states = hidden_states.sum(dim=1) / padding_mask.sum(dim=1).view(-1, 1)
+
+                    encoder_outputs = accelerator.gather_for_metrics(hidden_states)
+                    
+                    # TODO: check it works multi device
+                    if accelerator.is_main_process:
+                        all_encoder_outputs.extend(encoder_outputs.to("cpu").numpy())
+                        
+                if accelerator.is_main_process:
+                    tmp_hidden_states = Dataset.from_dict({"hidden_states": all_encoder_outputs})
+                    tmp_hidden_states.save_to_disk(os.path.join(data_args.temporary_save_to_disk, split))
+                accelerator.wait_for_everyone()
+                del all_encoder_outputs
+
+                tmp_hidden_states = datasets.load_from_disk(os.path.join(data_args.temporary_save_to_disk, split))
+                with accelerator.main_process_first():
+                    raw_datasets[split] = concatenate_datasets([raw_datasets[split].remove_columns("audio"), tmp_hidden_states], axis=1)
+                        
+            accelerator.free_memory()
+            del data_loader, batch, accelerator
+            
+        data_collator = DataCollatorHiddenStatesPadding()
+    else:
+        data_collator = DataCollatorFeatureExtractorWithPadding(feature_extractor, max_length_seconds=data_args.max_length_seconds)
+    
+    if data_args.save_to_disk is not None and not dataset_was_precomputed:
+        raw_datasets.save_to_disk(data_args.save_to_disk)
+        logger.info(f"Dataset saved at {data_args.save_to_disk}. Be careful of changing data parameters, which won't change the current saved dataset if reloaded.")
+
     # Initialize our trainer
     trainer = Trainer(
         model=model,
         args=training_args,
+        data_collator=data_collator,
         train_dataset=raw_datasets["train"] if training_args.do_train else None,
         eval_dataset=raw_datasets["eval"] if training_args.do_eval else None,
         compute_metrics=compute_metrics,
         tokenizer=feature_extractor,
     )
+    ignore_keys = ["hidden_states","attentions"] if model_args.use_weighted_layer_sum else None
 
     # Training
     if training_args.do_train:
@@ -945,7 +1221,7 @@ def main():
             checkpoint = training_args.resume_from_checkpoint
         elif last_checkpoint is not None:
             checkpoint = last_checkpoint
-        train_result = trainer.train(resume_from_checkpoint=checkpoint)
+        train_result = trainer.train(resume_from_checkpoint=checkpoint, ignore_keys_for_eval=ignore_keys)
         trainer.save_model()
         trainer.log_metrics("train", train_result.metrics)
         trainer.save_metrics("train", train_result.metrics)
@@ -953,7 +1229,7 @@ def main():
 
     # Evaluation
     if training_args.do_eval:
-        metrics = trainer.evaluate()
+        metrics = trainer.evaluate(ignore_keys=ignore_keys)
         trainer.log_metrics("eval", metrics)
         trainer.save_metrics("eval", metrics)