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wav2vec2_finetune_chunks.py
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wav2vec2_finetune_chunks.py
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import os
import random as random
from torch.utils.data import DataLoader
from data_utils.chunk_dataset import ChunkedDataset
os.environ["CUDA_VISIBLE_DEVICES"] = "4,5"
from dataclasses import dataclass
from glob import glob
from os import listdir, makedirs
from shutil import rmtree
from os.path import isfile, join
from typing import Dict, List, Optional, Tuple, Union
from sklearn.preprocessing import LabelEncoder
import sys
import librosa
import numpy as np
import pandas as pd
import datetime
import socket
import soundfile as sf
import torch
import torch.nn as nn
import torchaudio
import yaml
import evaluate
import torch.nn.functional as F
from datasets import Audio, Dataset, concatenate_datasets, load_dataset
from sklearn.metrics import accuracy_score, confusion_matrix, recall_score
from torch.nn import BCEWithLogitsLoss, CrossEntropyLoss, MSELoss
from transformers import (
AutoConfig,
AutoFeatureExtractor,
AutoModelForAudioClassification,
AutoTokenizer,
Trainer,
TrainingArguments,
Wav2Vec2FeatureExtractor,
Wav2Vec2Processor,
pipeline,
)
from transformers.file_utils import ModelOutput
from transformers.models.wav2vec2.modeling_wav2vec2 import (
Wav2Vec2Model,
Wav2Vec2PreTrainedModel,
)
@dataclass
class DataCollatorCTCWithPadding:
"""
Data collator that will dynamically pad the inputs received.
Args:
processor (:class:`~transformers.Wav2Vec2Processor`)
The processor used for proccessing the data.
padding (:obj:`bool`, :obj:`str` or :class:`~transformers.tokenization_utils_base.PaddingStrategy`, `optional`, defaults to :obj:`True`):
Select a strategy to pad the returned sequences (according to the model's padding side and padding index)
among:
* :obj:`True` or :obj:`'longest'`: Pad to the longest sequence in the batch (or no padding if only a single
sequence if provided).
* :obj:`'max_length'`: Pad to a maximum length specified with the argument :obj:`max_length` or to the
maximum acceptable input length for the model if that argument is not provided.
* :obj:`False` or :obj:`'do_not_pad'` (default): No padding (i.e., can output a batch with sequences of
different lengths).
max_length (:obj:`int`, `optional`):
Maximum length of the ``input_values`` of the returned list and optionally padding length (see above).
max_length_labels (:obj:`int`, `optional`):
Maximum length of the ``labels`` returned list and optionally padding length (see above).
pad_to_multiple_of (:obj:`int`, `optional`):
If set will pad the sequence to a multiple of the provided value.
This is especially useful to enable the use of Tensor Cores on NVIDIA hardware with compute capability >=
7.5 (Volta).
"""
processor: Wav2Vec2Processor
padding: Union[bool, str] = True
max_length: Optional[int] = None
max_length_labels: Optional[int] = None
pad_to_multiple_of: Optional[int] = None
pad_to_multiple_of_labels: Optional[int] = None
def __call__(
self, features: List[Dict[str, Union[List[int], torch.Tensor]]]
) -> Dict[str, torch.Tensor]:
input_features = [
{"input_values": feature["input_values"]} for feature in features
]
# d_type = torch.long if isinstance(label_features[0], int) else torch.float
d_type = torch.long
batch = self.processor.pad(
input_features,
padding=self.padding,
max_length=self.max_length,
truncation=True,
pad_to_multiple_of=self.pad_to_multiple_of,
return_tensors="pt",
)
if "labels" in features[0]:
label_features = [feature["labels"] for feature in features]
batch["labels"] = torch.tensor(label_features, dtype=d_type)
return batch
def speech_file_to_array_fn(path):
speech_array, sampling_rate = torchaudio.load(path)
resampler = torchaudio.transforms.Resample(sampling_rate, target_sampling_rate)
speech = resampler(speech_array).squeeze().numpy()
return speech
def speech_file_to_array_chunks(start, stop, path):
utterance, _ = sf.read(path, dtype="float32", start=start, stop=stop)
# Convert to torch tensor
# utterance_torch = torch.from_numpy(utterance)
return utterance
def get_random_chunks(length, segment):
seg_len = int(segment * 16000)
start = random.randint(0, length - seg_len)
stop = start + seg_len
return start, stop
def preprocess_function(examples):
segments = [get_random_chunks(length, 2) for length in examples["length"]]
speech_list = [speech_file_to_array_chunks(segments[0][0], segments[0][1], path) for path in examples["path"]]
result = processor(speech_list, sampling_rate=target_sampling_rate)
if "label" in examples:
target_list = [
int(label) for label in examples["label"]
] # Do any preprocessing on your float/integer data
result["labels"] = list(target_list)
return result
@dataclass
class SpeechClassifierOutput(ModelOutput):
loss: Optional[torch.FloatTensor] = None
logits: torch.FloatTensor = None
hidden_states: Optional[Tuple[torch.FloatTensor]] = None
attentions: Optional[Tuple[torch.FloatTensor]] = None
class Wav2Vec2ClassificationHead(nn.Module):
"""Head for wav2vec classification task."""
def __init__(self, config):
super().__init__()
self.dense = nn.Linear(config.hidden_size, config.hidden_size)
self.dropout = nn.Dropout(config.final_dropout)
self.out_proj = nn.Linear(config.hidden_size, config.num_labels)
def forward(self, features, **kwargs):
x = features
x = self.dropout(x)
x = self.dense(x)
x = torch.tanh(x)
x = self.dropout(x)
x = self.out_proj(x)
return x
class Wav2Vec2ForSpeechClassification(Wav2Vec2PreTrainedModel):
def __init__(self, config):
super().__init__(config)
self.num_labels = config.num_labels
self.pooling_mode = config.pooling_mode
self.config = config
self.wav2vec2 = Wav2Vec2Model(config)
self.classifier = Wav2Vec2ClassificationHead(config)
self.init_weights()
def freeze_feature_extractor(self):
self.wav2vec2.feature_extractor._freeze_parameters()
def merged_strategy(self, hidden_states, mode="mean"):
if mode == "mean":
outputs = torch.mean(hidden_states, dim=1)
elif mode == "sum":
outputs = torch.sum(hidden_states, dim=1)
elif mode == "max":
outputs = torch.max(hidden_states, dim=1)[0]
else:
raise Exception(
"The pooling method hasn't been defined! Your pooling mode must be one of these ['mean', 'sum', 'max']"
)
return outputs
def forward(
self,
input_values,
attention_mask=None,
output_attentions=None,
output_hidden_states=None,
return_dict=None,
labels=None,
):
return_dict = (
return_dict if return_dict is not None else self.config.use_return_dict
)
outputs = self.wav2vec2(
input_values,
attention_mask=attention_mask,
output_attentions=output_attentions,
output_hidden_states=output_hidden_states,
return_dict=return_dict,
)
hidden_states = outputs[0]
hidden_states = self.merged_strategy(hidden_states, mode=self.pooling_mode)
logits = self.classifier(hidden_states)
loss = None
if labels is not None:
if self.config.problem_type is None:
if self.num_labels == 1:
self.config.problem_type = "regression"
elif self.num_labels > 1 and (
labels.dtype == torch.long or labels.dtype == torch.int
):
self.config.problem_type = "single_label_classification"
else:
self.config.problem_type = "multi_label_classification"
if self.config.problem_type == "regression":
loss_fct = F.mse_loss
loss = loss_fct(logits.view(-1, self.num_labels), labels.unsqueeze(-1))
elif self.config.problem_type == "single_label_classification":
loss_fct = CrossEntropyLoss()
loss = loss_fct(logits.view(-1, self.num_labels), labels.view(-1))
elif self.config.problem_type == "multi_label_classification":
loss_fct = BCEWithLogitsLoss()
loss = loss_fct(logits, labels)
if not return_dict:
output = (logits,) + outputs[2:]
return ((loss,) + output) if loss is not None else output
return SpeechClassifierOutput(
loss=loss,
logits=logits,
hidden_states=outputs.hidden_states,
attentions=outputs.attentions,
)
def compute_metrics(eval_pred):
"""Computes spearman and pearson on predictions"""
recall = recall_metric.compute(
predictions=np.argmax(eval_pred.predictions.squeeze(), axis=-1),
references=eval_pred.label_ids,
average="macro",
)
f1 = f1_metric.compute(
predictions=np.argmax(eval_pred.predictions.squeeze(), axis=-1),
references=eval_pred.label_ids,
average="macro",
)
# f1 = f1_metric.compute(predictions=predictions, references=eval_pred.label_ids, average="macro")
return {**recall, **f1}
if __name__ == "__main__":
with open("config/params.yaml") as f:
_params = yaml.safe_load(f)
params = _params["wav2vec"]
target = _params["target"]
model_used = _params['wav2vec']["model"].split("/")[-1]
label_dir = "data/eating"
# result_dir = "results/{}".format(model_used)
result_dir = "results/chunks_{}".format(model_used)
logging_dir = "tb_logs/{}".format(model_used)
model_checkpoint = params["model"]
batch_size = 6
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
target_sr = 16000
model_name = model_checkpoint.split("/")[-1]
all_files = list(glob("./data/wav/*.wav", recursive=True))
label_list = [0]
makedirs(result_dir, exist_ok=True)
makedirs(logging_dir, exist_ok=True)
recall_metric = evaluate.load("recall")
f1_metric = evaluate.load("f1")
df_train = pd.read_csv(f"{label_dir}/train_chunked_2secs.csv", encoding="utf-8")
df_dev = pd.read_csv(f"{label_dir}/dev_chunked_2secs.csv", encoding="utf-8")
df_test = pd.read_csv(f"{label_dir}/test_chunked_2secs.csv", encoding="utf-8")
# df_train["filename_full"] = "/srv/data/egasj/corpora/eating-wav-all/" + df_train["filename"] + ".wav"
# df_dev["filename_full"] = "/srv/data/egasj/corpora/eating-wav-all/" + df_dev["filename"] + ".wav"
# df_test["filename_full"] = "/srv/data/egasj/corpora/eating-wav-all/" + df_test["filename"] + ".wav"
label_encoder = LabelEncoder()
df_train["label"] = label_encoder.fit_transform(df_train['label'])
df_dev["label"] = label_encoder.transform(df_dev['label'])
# if len(set(df_test[target])) > 1:
df_test["label"] = label_encoder.transform(df_test['label'])
train_dataset = Dataset.from_pandas(df_train)
dev_dataset = Dataset.from_pandas(df_dev)
test_dataset = Dataset.from_pandas(df_test)
config = AutoConfig.from_pretrained(
model_checkpoint,
num_labels=len(label_encoder.classes_),
problem_type=None,
)
setattr(config, "pooling_mode", params["pooling"])
processor = AutoFeatureExtractor.from_pretrained(model_checkpoint)
target_sampling_rate = processor.sampling_rate
data_collator = DataCollatorCTCWithPadding(
processor=processor, padding=True, max_length=10 * target_sampling_rate
)
train_dataset = train_dataset.map(
preprocess_function, batch_size=100, batched=True, num_proc=4
)
print("processed train")
dev_dataset = dev_dataset.map(
preprocess_function, batch_size=100, batched=True, num_proc=4
)
print("processed devel")
test_dataset = test_dataset.map(
preprocess_function, batch_size=32, batched=True, num_proc=4
)
print("processed test")
processor = Wav2Vec2Processor.from_pretrained(model_checkpoint)
target_sampling_rate = processor.feature_extractor.sampling_rate
data_collator = DataCollatorCTCWithPadding(
processor=processor, padding=True, max_length=10 * target_sampling_rate
)
model = Wav2Vec2ForSpeechClassification.from_pretrained(
model_checkpoint,
config=config,
)
if params["freezeExtractor"]:
model.freeze_feature_extractor()
if params["freezeTransformer"]:
model.freeze_transformer()
args = TrainingArguments(
result_dir,
evaluation_strategy="epoch",
save_strategy="epoch",
learning_rate=3e-4,
per_device_train_batch_size=batch_size,
gradient_accumulation_steps=1,
per_device_eval_batch_size=batch_size,
num_train_epochs=params["epochs"],
warmup_ratio=0.1,
logging_steps=20,
fp16=True,
load_best_model_at_end=True,
metric_for_best_model="recall",
push_to_hub=False,
gradient_checkpointing=True,
save_total_limit=2,
disable_tqdm=False,
logging_dir=f"{logging_dir}/{datetime.datetime.now().strftime('%h%d_%Y-%H-%M-%S')}_{socket.gethostname()}",
)
print(args.device)
trainer = Trainer(
model=model,
args=args,
data_collator=data_collator,
train_dataset=train_dataset,
eval_dataset=dev_dataset,
tokenizer=processor.feature_extractor,
compute_metrics=compute_metrics,
)
trainer.train()
dev_predictions = trainer.predict(dev_dataset)
dev_df = pd.DataFrame(
{
"filename": dev_dataset["filename"],
"prediction": label_encoder.inverse_transform(
np.argmax(dev_predictions.predictions.squeeze(), axis=-1)
),
"true": label_encoder.inverse_transform(
dev_predictions.label_ids.squeeze()
),
}
)
dev_df.to_csv(join(result_dir, "predictions.devel.csv"), index=False)
print("DEV -->", compute_metrics(dev_predictions))
test_predictions = trainer.predict(test_dataset)
test_df = pd.DataFrame(
{
"filename": test_dataset["filename"],
"prediction": label_encoder.inverse_transform(
np.argmax(test_predictions.predictions.squeeze(), axis=-1)
),
"true": label_encoder.inverse_transform(
test_predictions.label_ids.squeeze()
)
# if len(set(test_predictions.label_ids)) > 1
# else ["?"] * test_predictions.predictions.shape[0],
}
)
test_df.to_csv(join(result_dir, "predictions.test.csv"), index=False)
print("TEST-->",compute_metrics(test_predictions))
# DEV --> {'recall': 0.761, 'f1': 0.754} --> 2 secs chunks
# TEST--> {'recall': 0.793, 'f1': 0.787} --> 2 secs chunks
# DEV --> {'recall': 0., 'f1': 0.} --> 2 secs chunks + handle remaining chunks
# TEST--> {'recall': 0., 'f1': 0.} --> 2 secs chunks