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stt3_train.py
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stt3_train.py
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#!/usr/bin/env python3
from re import A
from shutil import ExecError
import numpy as np
import torch
import torch.nn as nn
import time
from packaging import version
import math
from dataloader_np import *
from glob import glob
import os
import pandas as pd
import pickle
from datasets import load_metric
import argparse
import nsml
from nsml import HAS_DATASET, DATASET_PATH
from torch.optim.lr_scheduler import LambdaLR
from tqdm import tqdm
import json
from pydub import AudioSegment
from pydub.silence import split_on_silence
from transformers import Wav2Vec2ForCTC, Wav2Vec2Config
from apex import amp
from ctcdecode import CTCBeamDecoder
def evaluate(model, batch,tokenizer, beam_decoder):
model.eval()
with torch.no_grad():
logits = model(batch).logits
beam_results, beam_scores, timesteps, out_lens = beam_decoder.decode(logits)
result_list = []
for token,out_len in zip(beam_results.cpu().numpy(),out_lens):
a = tokenizer.convert(token[0][:out_len[0]],predicted=False)
result_list.append(a)
return result_list
def split_to_chunk(speech,result_chunk,addition_flag,min_silence_len):
if min_silence_len < 200:
return
audio_chunks = split_on_silence(speech, min_silence_len=min_silence_len, silence_thresh=-40)
for i,chunk in enumerate(audio_chunks):
if chunk.frame_count() > 200000:
split_to_chunk(chunk,result_chunk, addition_flag, min_silence_len-100)
else:
result_chunk.append(chunk)
if chunk.frame_count() < 16000:
addition_flag.append(1)
else:
addition_flag.append(0)
return
def save_checkpoint(checkpoint, dir):
torch.save(checkpoint, os.path.join(dir))
def bind_model(model, parser):
# 학습한 모델을 저장하는 함수입니다.
def save(dir_name, *parser):
# directory
os.makedirs(dir_name, exist_ok=True)
save_dir = os.path.join(dir_name, "checkpoint")
save_checkpoint(dict_for_infer, save_dir)
with open(os.path.join(dir_name, "dict_for_infer"), "wb") as f:
pickle.dump(dict_for_infer, f)
print("저장 완료!")
# 저장한 모델을 불러올 수 있는 함수입니다.
def load(dir_name, *parser):
save_dir = os.path.join(dir_name, "checkpoint")
global checkpoint
checkpoint = torch.load(save_dir)
global dict_for_infer
with open(os.path.join(dir_name, "dict_for_infer"), "rb") as f:
dict_for_infer = pickle.load(f)
tokenizer = dict_for_infer["tokenizer"]
model.lm_head = nn.Linear(
in_features=768, out_features=len(tokenizer.txt2idx), bias=True
)
model.config = Wav2Vec2Config(vocab_size=len(tokenizer.txt2idx))
#model.load_state_dict(checkpoint["model"])
print("로딩 완료!")
def infer(test_path, **kwparser):
device = checkpoint["device"]
test_file_list = sorted(
glob(os.path.join(DATASET_PATH, "test", "test_data", "*"))
)
files = []
for file_num, file in enumerate(test_file_list):
addition_flag = []
result_chunk = []
sound = AudioSegment.from_wav(file)
sound = match_target_amplitude(sound, -20.0)
split_to_chunk(sound,result_chunk,addition_flag,500)
result = []
for i, flag in enumerate(addition_flag):
if flag == 1 and i < len(addition_flag) -1:
result_chunk[i+1] = result_chunk[i] + result_chunk[i + 1]
elif flag == 0 or result_chunk[i].frame_count() > 80000:
result.append(result_chunk[i])
elif i == len(addition_flag) -1:
result[-1] = result[-1] + result_chunk[i]
for j, chunk in enumerate(result):
new_file = str(file_num) + '_' + str(j)
np.save(new_file, chunk.get_array_of_samples())
files.append(new_file + '.npy')
test_data = pd.DataFrame()
test_data['file'] = pd.Series(files)
test_dataset = CustomDataset(test_data, mode="test")
test_sampler = RandomBucketBatchSampler(
test_dataset, batch_size=dict_for_infer["batch_size"], drop_last=False
)
callate_fn = AudioCollate()
test_data_loader = DataLoader(
test_dataset,batch_size=dict_for_infer["batch_size"], collate_fn=callate_fn, num_workers=8,pin_memory=True
)
tokenizer = dict_for_infer["tokenizer"]
beam_decoder = CTCBeamDecoder(tokenizer.vocab,
#model_path='n-gram/stt2_n2.binary',
alpha=0, beta=0,
cutoff_top_n=20, cutoff_prob=1.0,
beam_width=100, num_processes=4,
blank_id=tokenizer.txt2idx["<pad>"],
log_probs_input=True)
model.to(device)
if args.fp16 and args.mode == 'test':
model2 = amp.initialize(model, opt_level="O1")
print('fp16 on')
else:
model2 = model
model2.load_state_dict(checkpoint["model"])
result_list = []
result_file_list = []
for step, batch in enumerate(test_data_loader):
speech = batch["speech"].to(device)
output = evaluate(model2, speech,tokenizer, beam_decoder)
result_list.extend(output)
result_file_list.extend(batch['file'])
final_result = [str() for i in range(len(test_file_list))]
for r,f in zip(result_list,result_file_list):
num = int(f.split('_')[0])
final_result[num] = final_result[num] + r + ' '
for i in range(len(final_result)):
final_result[i] = final_result[i].strip()
files = []
for file in test_file_list:
files.append(file.split('/')[-1])
# DONOTCHANGE: They are reserved for nsml
# 리턴 결과는 [(확률, 0 or 1)] 의 형태로 보내야만 리더보드에 올릴 수 있습니다. 리더보드 결과에 확률의 값은 영향을 미치지 않습니다
# return list(zip(pred.flatten(), clipped.flatten()))
return list(zip(files, final_result))
# DONOTCHANGE: They are reserved for nsml
# nsml에서 지정한 함수에 접근할 수 있도록 하는 함수입니다.
nsml.bind(save=save, load=load, infer=infer)
def get_cosine_schedule_with_warmup(
optimizer, num_warmup_steps, num_training_steps, num_cycles=0.5, last_epoch=-1
):
"""Create a schedule with a learning rate that decreases following the
values of the cosine function between 0 and `pi * cycles` after a warmup
period during which it increases linearly between 0 and 1.
"""
def lr_lambda(current_step):
if current_step < num_warmup_steps:
return float(current_step) / float(max(1, num_warmup_steps))
progress = float(current_step - num_warmup_steps) / float(
max(1, num_training_steps - num_warmup_steps)
)
return max(
0.0, 0.5 * (1.0 + math.cos(math.pi * float(num_cycles) * 2.0 * progress))
)
return LambdaLR(optimizer, lr_lambda, last_epoch)
def validate(valid_dataloader, model, tokenizer):
model.eval()
device = next(model.parameters()).device
metric = load_metric("cer")
total = len(valid_dataloader)
alpha=0
beta=0
beam_width = 100
beam_decoder = CTCBeamDecoder(tokenizer.vocab,
alpha=alpha, beta=beta,
cutoff_top_n=40, cutoff_prob=1.0,
beam_width=beam_width, num_processes=11,
blank_id=tokenizer.txt2idx["<pad>"],
log_probs_input=True)
for i, batch in enumerate(tqdm(valid_dataloader)):
print("validation:" + str(i) + "/" + str(total))
with torch.no_grad():
speech = batch["speech"].to(device)
text = batch["labels"].to(device)
print(speech.shape)
model_predictions = model(speech, labels=text).logits
'''predicted_ids = torch.argmax(model_predictions, dim=-1)
predictions = [
tokenizer.convert(sen) for sen in predicted_ids.cpu().numpy()
]'''
beam_results, beam_scores, timesteps, out_lens = beam_decoder.decode(model_predictions)
result_list = []
for token,out_len in zip(beam_results.cpu().numpy(),out_lens):
#result_list.append("".join([tokenizer.idx2txt[x] for x in token]))
a = tokenizer.convert(token[0][:out_len[0]],predicted=False)
result_list.append(a)
references = [tokenizer.convert(sen,predicted=False) for sen in text.cpu().numpy()]
print(result_list[0])
metric.add_batch(predictions=result_list, references=references)
final_score = metric.compute()
print(final_score)
return {"cer": final_score}
def clean(sen):
cleaned_sen = re.sub('SP|FP|SN|NO|\(|\)|:|\*|,|…|\{[^\}]+\}','',sen)
cleaned_sen = re.sub('&[^&]+&','m',cleaned_sen)
cleaned_sen = re.sub('\s{2,}',' ',cleaned_sen)
return cleaned_sen
def match_target_amplitude(sound, target_dBFS):
change_in_dBFS = target_dBFS - sound.dBFS
return sound.apply_gain(change_in_dBFS)
if __name__ == "__main__":
# See all possible arguments in src/transformers/training_args.py
# or by passing the --help flag to this script.
# We now keep distinct sets of args, for a cleaner separation of concerns.
parser = argparse.ArgumentParser()
parser.add_argument("--mode", type=str, default="train")
parser.add_argument("--iteration", type=str, default="0")
parser.add_argument("--pause", type=int, default=0)
parser.add_argument("--seed", type=int, default=40)
parser.add_argument("--batch_size", type=int, default=20)
parser.add_argument("--fp16", action="store_true")
parser.add_argument("--total_epoch", type=int, default=200)
parser.add_argument("--warmup_step", type=int, default=15000)
parser.add_argument("--lr", type=float, default=5e-5)
parser.add_argument("--log_every", type=int, default=1)
parser.add_argument("--valid_every", type=int, default=5000)
parser.add_argument("--save_every", type=int, default=5000)
parser.add_argument("--strategy", type=str, default="step")
parser.add_argument("--max_vocab_size", type=int, default=-1)
parser.add_argument("--reload_from", type=int, default=0)
parser.add_argument("--checkpoint", type=str)
parser.add_argument("--session", type=str)
args = parser.parse_args()
global dict_for_infer
model = Wav2Vec2ForCTC.from_pretrained("nuod/wav2vec2")
model.freeze_feature_extractor()
bind_model(model=model, parser=args)
if args.pause:
nsml.paused(scope=locals())
if args.mode == "train":
train_path = os.path.join(DATASET_PATH, "train")
wav_path = os.path.join(train_path, "train_data","wav")
json_list = sorted(glob(os.path.join(train_path, "train_data","train_info", "*")))
label = pd.read_csv(os.path.join(train_path, "train_label"))
split_num = int(len(label) * 0.995)
train_label = label.iloc[:split_num]
val_label = label.iloc[split_num:]
splited_train_labels = []
splited_valid_labels = []
splited_train_file = []
splited_valid_file = []
train_duration = []
valid_duration = []
file_cnt = 0
print('making np_data')
print(len(json_list))
for k, file in enumerate(json_list):
print(k)
with open(file,'r',encoding='UTF8') as f:
data = json.load(f)
#sound, sr = librosa.load(os.path.join(wav_path,data['id']), sr=16000)
#sound = match_target_amplitude(sound, -20.0).get_array_of_samples()
sound = AudioSegment.from_wav(os.path.join(wav_path,data['id']))
sr = 16000
sound = sound.set_frame_rate(16000)
print(sound.frame_rate)
sound = sound.get_array_of_samples()
for value in data['utterance']:
np_sound = sound[int(value['start'] * sr): int(value['end'] * sr)]
if data['id'] in val_label['file_name'].values:
splited_valid_labels.append(value['dialect_form'])
file_name = str(file_cnt)
splited_valid_file.append(file_name + '.npy')
valid_duration.append(len(np_sound))
else:
splited_train_labels.append(value['dialect_form'])
file_name = str(file_cnt)
splited_train_file.append(file_name + '.npy')
train_duration.append(len(np_sound))
file_cnt += 1
np.save(file_name,np_sound)
print('done')
train_data = pd.DataFrame()
train_data['file'] = pd.Series(splited_train_file)
train_data['target'] = pd.Series(splited_train_labels)
train_data['length'] = pd.Series(train_duration)
valid_data = pd.DataFrame()
valid_data['file'] = pd.Series(splited_valid_file)
valid_data['target'] = pd.Series(splited_valid_labels)
valid_data['length'] = pd.Series(valid_duration)
train_label = [clean(sen) for sen in train_data.target]
val_label = [clean(sen) for sen in valid_data.target]
if args.reload_from != 0:
nsml.load(args.checkpoint, session = args.session)
tokenizer = dict_for_infer['tokenizer']
args.batch_size = dict_for_infer['batch_size']
args.total_epoch = dict_for_infer['epochs']
args.lr = dict_for_infer['learning_rate']
else:
tokenizer = CustomTokenizer()
tokenizer.fit(train_label)
print(tokenizer.txt2idx)
model.lm_head = nn.Linear(
in_features=768, out_features=len(tokenizer.txt2idx), bias=True
)
model.config = Wav2Vec2Config(vocab_size=len(tokenizer.txt2idx))
train_tokens = tokenizer.txt2token(train_label)
valid_tokens = tokenizer.txt2token(val_label)
train_data['target'] = pd.Series(train_tokens)
valid_data['target'] = pd.Series(valid_tokens)
train_dataset = CustomDataset(train_data, max_size=200000, min_size=5000)
valid_dataset = CustomDataset(valid_data, max_size=200000, min_size=5000)
print(len(train_dataset))
print('-'*80)
train_batch_sampler = RandomBucketBatchSampler(
train_dataset, batch_size=args.batch_size, drop_last=False
)
valid_batch_sampler = RandomBucketBatchSampler(
valid_dataset, batch_size=args.batch_size, drop_last=False
)
collate_fn = TextAudioCollate()
train_dataloader = DataLoader(
train_dataset,
batch_sampler=train_batch_sampler,
collate_fn=collate_fn,
num_workers=11,
pin_memory=True
)
valid_dataloader = DataLoader(
valid_dataset,
batch_sampler=valid_batch_sampler,
collate_fn=collate_fn,
num_workers=11,
pin_memory=True
)
np.random.seed(args.seed)
torch.manual_seed(args.seed)
torch.cuda.manual_seed(args.seed)
device = torch.device(f"cuda:0" if torch.cuda.is_available() else "cpu")
###############################################################################
# Prepare the Optimizer
###############################################################################
no_decay = ["bias", "LayerNorm.weight"]
optimizer_grouped_parameters = [
{
"params": [
p
for n, p in model.named_parameters()
if not any(nd in n for nd in no_decay)
],
"weight_decay": 0.01,
},
{
"params": [
p
for n, p in model.named_parameters()
if any(nd in n for nd in no_decay)
],
"weight_decay": 0.0,
},
]
optimizer = torch.optim.AdamW(optimizer_grouped_parameters, lr=args.lr)
scheduler = get_cosine_schedule_with_warmup(
optimizer,
num_warmup_steps=args.warmup_step,
num_training_steps=len(train_dataloader) * args.total_epoch,
) # do not foget to modify the number when dataset is changed
model.to(device)
if args.fp16:
try:
from apex import amp
except ImportError:
raise ImportError(
"Please install apex from https://www.github.com/nvidia/apex to use fp16 training."
)
model, optimizer = amp.initialize(model, optimizer, opt_level="O1")
if args.reload_from != 0:
optimizer.load_state_dict(dict_for_infer['opt'])
model.load_state_dict(dict_for_infer['model'])
scheduler.load_state_dict(dict_for_infer['scaler'])
if args.fp16:
amp.load_state_dict(checkpoint['amp'])
n_iters = len(train_dataloader)
if args.strategy == "epoch":
args.valid_every = n_iters
args.save_every = n_iters
itr_global = args.reload_from + 1
for epoch in range(int(args.reload_from / n_iters) + 1, args.total_epoch + 1):
itr_start_time = time.time()
losses = []
for batch in train_dataloader:
model.train()
speech = batch["speech"].to(device)
text = batch["labels"].to(device)
loss = model(speech, labels=text).loss
if args.fp16:
with amp.scale_loss(loss, optimizer) as scaled_loss:
scaled_loss.backward()
torch.nn.utils.clip_grad_norm_(amp.master_params(optimizer), 5.0)
else:
loss.backward()
torch.nn.utils.clip_grad_norm_(model.parameters(), 5.0)
optimizer.step()
scheduler.step()
model.zero_grad()
losses.append(loss.item())
if itr_global % args.log_every == 0:
elapsed = time.time() - itr_start_time
print(
"epo:[%d/%d] itr:[%d/%d] step_time:%ds Loss=%.5f"
% (
epoch,
args.total_epoch,
itr_global % n_iters,
n_iters,
elapsed,
np.mean(losses),
)
)
summary = {"summary": True, "scope": locals(), "step": itr_global}
summary.update({"loss": np.mean(losses)})
nsml.report(**summary)
losses = []
itr_start_time = time.time()
itr_global = itr_global + 1
if itr_global % args.valid_every == 0:
print("validating..")
valid_result = validate(valid_dataloader, model, tokenizer)
print(valid_result)
summary = {"summary": True, "scope": locals(), "step": itr_global}
summary.update(valid_result)
nsml.report(**summary)
dict_for_infer = {
"model": model.state_dict(),
"opt": optimizer.state_dict(),
"scaler": scheduler.state_dict(),
"amp": amp.state_dict(),
"batch_size": args.batch_size,
"epochs": args.total_epoch,
"learning_rate": args.lr,
"tokenizer": tokenizer,
"device": device,
}
if itr_global % args.save_every == 0:
print("saving...")
nsml.save(checkpoint=f"{itr_global}")