import argparse
def parse_args():
parser = argparse.ArgumentParser(description="encode the librilight dataset using encodec model")
parser.add_argument("--dataset_size", type=str, default='xs', help='sizes of gigaspeech, xs, s, m, l, xl. we use xl for VoiceCraft training, xs is good for debugging')
parser.add_argument('--download_to', type=str, default="/data/scratch/pyp/datasets/gigaspeech_debug", help="dir where you want the huggingface gigaspeech dataset to be downloaded to")
parser.add_argument('--save_dir', type=str, default="/data/scratch/pyp/datasets/gigaspeech_phn_enc_manifest_debug", help="path to the manifest, phonemes, and encodec codes dirs")
parser.add_argument('--encodec_model_path', type=str, default="/data/scratch/pyp/exp_pyp/audiocraft/encodec/xps/6f79c6a8/checkpoint.th")
parser.add_argument('--n_workers', type=int, default=4, help="Number of parallel worker processes")
parser.add_argument('--mega_batch_size', type=int, default=100, help="Number of samples in each mega batch for multiprocess dataloading")
parser.add_argument('--batch_size', type=int, default=4, help="batch size for encodec encoding, decrease it if OOM. This is the sum of batch size *over each gpu*, so increase it if you are using more gpus")
parser.add_argument('--model_sr', type=int, default=16000, help='encodec input audio sample rate')
parser.add_argument('--downsample_rate', type=int, default=320, help='encodec downsample rate')
parser.add_argument('--model_code_sr', type=int, default=50, help='encodec model code sample rate')
parser.add_argument('--len_cap', type=float, default=35.0, help='will drop audios that are longer than this number')
parser.add_argument('--max_len', type=int, default=30000, help='max length of audio in samples, if exceed, will cut a batch into half to process, decrease this number if OOM on your machine')
return parser.parse_args()
if __name__ == "__main__":
import logging
formatter = (
"%(asctime)s [%(levelname)s] %(filename)s:%(lineno)d || %(message)s"
)
logging.basicConfig(format=formatter, level=logging.INFO)
args = parse_args()
import os
import numpy as np
import torch
import tqdm
import time
from datasets import load_dataset, DownloadConfig
from tokenizer import TextTokenizer, tokenize_text
# get the path
phn_save_root = os.path.join(args.save_dir, args.dataset_size, "phonemes")
codes_save_root = os.path.join(args.save_dir, args.dataset_size, "encodec_16khz_4codebooks")
vocab_fn = os.path.join(args.save_dir, args.dataset_size, "vocab.txt")
os.makedirs(phn_save_root, exist_ok=True)
os.makedirs(codes_save_root, exist_ok=True)
def sort_by_audio_len(lens):
inds = np.argsort(lens).tolist()
logging.info(f"longest: {lens[inds[-1]]*args.model_code_sr} encodec codes, {lens[inds[-1]]:.2f} sec.")
logging.info(f"shortest: {lens[inds[0]]*args.model_code_sr} encodec codes, {lens[inds[0]]:.2f} sec.")
logging.info(f"median: {lens[inds[len(inds)//2]]*args.model_code_sr} encodec codes, {lens[inds[len(inds)//2]]:.2f} sec.")
logging.info(f"95 percentile longest: {lens[inds[int(len(inds)*0.95)]]*args.model_code_sr} encodec codes, {lens[inds[int(len(inds)*0.95)]]:.2f} sec.")
return inds[::-1]
def write_array_to_txt_file(array, filename):
with open(filename, 'w') as f:
for a in array[:-1]:
f.write(' '.join(map(str, a))+'\n')
f.write(' '.join(map(str, array[-1])))
### phonemization
# load tokenizer
# load the encodec model
from audiocraft.solvers import CompressionSolver
model = CompressionSolver.model_from_checkpoint(args.encodec_model_path)
model = model.cuda()
model = model.eval()
text_tokenizer = TextTokenizer()
# https://github.com/SpeechColab/GigaSpeech
# there are only four different punctuations
# need to check whether there are other < started strings
punc2sym = {" <COMMA>": ",", " <PERIOD>": ".", " <QUESTIONMARK>": "?", " <EXCLAMATIONPOINT>": "!"} # note the space in front of each punc name
gar2sym = {"<SIL>": "#%#", "<MUSIC>": "##%", "<NOISE>": "%%#", "<OTHER>":"%#%"} # so that they are savely keep as the original sym when using tokenize_text
punc2sym.update(gar2sym)
word2sym = { "h æ ʃ h ɐ ʃ p ɚ s ɛ n t": "<MUSIC>", "h æ ʃ p ɚ s ɛ n t h æ ʃ": "<SIL>", "p ɚ s ɛ n t h ɐ ʃ p ɚ s ɛ n t": "<OTHER>", "p ɚ s ɛ n t p ɚ s ɛ n t h æ ʃ": "<NOISE>"}
forbidden_words = set(['#%#', '##%', '%%#', '%#%'])
dc = DownloadConfig(cache_dir=args.download_to)
stime = time.time()
logging.info("loading the dataset...")
gs = load_dataset("speechcolab/gigaspeech", args.dataset_size, use_auth_token=True, cache_dir = args.download_to, download_config=dc)
logging.info(f"time spend on loading the dataset: {time.time() - stime:.2f} seconds")
splits = ['validation', 'test', 'train']
logging.info(f"gigaspeech dataset {args.dataset_size} info: {gs}")
logging.info(f"phonemizing...")
phn_vocab = set()
all_lens = []
# you will see a ton of [WARNING] words_mismatch.py:88......, it's not a issue
for split in tqdm.tqdm(splits):
skip = 0
logging.info(f"now processing split {split}...")
for item in tqdm.tqdm(gs[split]):
save_fn = os.path.join(phn_save_root, item['segment_id']+".txt")
text = item['text']
if sum(word in forbidden_words for word in text.split(" ")):
logging.info(f"skip {item['segment_id']}, because it contains forbiden words. It's transcript: {text}")
skip += 1
continue
for k, v in punc2sym.items():
text = text.replace(k, v)
phn = tokenize_text(text_tokenizer, text)
phn_seq = " ".join(phn)
for k, v in word2sym.items():
phn_seq = phn_seq.replace(k, v)
phn_vocab.update(phn_seq.split(" "))
all_lens.append(len(phn_seq.split(" ")))
with open(save_fn, "w") as f:
f.write(phn_seq)
logging.info(f"split {split} has {len(gs[split])} samples in total, skipped {skip} due to forbiden words")
print(f"phn vocab size: {len(list(phn_vocab))}")
print("phn sequence stats: ")
print(f"longest: {max(all_lens)}")
print(f"shortest: {min(all_lens)}")
print(f"median: {np.quantile(all_lens, 0.5)}")
print(f"95 percentile longest: {np.quantile(all_lens, 0.95)}")
print("write vocabulary to ", vocab_fn)
with open(vocab_fn, "w") as f:
for i, phn in enumerate(list(phn_vocab)):
if i < len(list(phn_vocab)) - 1:
f.write(f"{str(i)} {phn}\n")
else:
f.write(f"{str(i)} {phn}")
class mydataset(torch.utils.data.Dataset):
def __init__(self, split):
super().__init__()
self.data = gs[split]
def __len__(self):
return len(self.data)
def __getitem__(self, ind):
try:
segment_id, audio, sr, text, begin_time, end_time = self.data[ind]['segment_id'], torch.from_numpy(self.data[ind]['audio']['array']).float(), self.data[ind]['audio']['sampling_rate'], self.data[ind]['text'], self.data[ind]['begin_time'], self.data[ind]['end_time']
except:
return None, None, None, None, None, None
return segment_id, audio, sr, text, begin_time, end_time
def collate(self, batch):
res = {'segment_id': [], "audio": [], "sr": [], "text": [], "begin_time": [], "end_time": []}
for item in batch:
if item[0] != None:
res['segment_id'].append(item[0])
res['audio'].append(item[1])
res['sr'].append(item[2])
res['text'].append(item[3])
res['begin_time'].append(item[4])
res['end_time'].append(item[5])
return res
## encodec codes extraction
logging.info("encodec encoding...")
train_dataset = mydataset('train')
train_loader = torch.torch.utils.data.DataLoader(train_dataset, batch_size=args.mega_batch_size, shuffle=False, drop_last=False, num_workers=args.n_workers, collate_fn=train_dataset.collate)
validation_dataset = mydataset('validation')
validation_loader = torch.torch.utils.data.DataLoader(validation_dataset, batch_size=args.mega_batch_size, shuffle=False, drop_last=False, num_workers=args.n_workers, collate_fn=validation_dataset.collate)
test_dataset = mydataset('test')
test_loader = torch.torch.utils.data.DataLoader(test_dataset, batch_size=args.mega_batch_size, shuffle=False, drop_last=False, num_workers=args.n_workers, collate_fn=test_dataset.collate)
splits = ['validation', 'test', 'train']
loaders = [validation_loader, test_loader, train_loader]
# splits = ['validation'] # for debug
# loaders = [validation_loader]
for split, loader in zip(splits, loaders):
skip = 0
logging.info(f"now processing split {split}...")
mega_n_steps = int(np.ceil(len(gs[split]) / args.mega_batch_size))
logging.info(f"partition the split {split} into {mega_n_steps} parts, each has {args.mega_batch_size} samples")
for m, mega_batch in enumerate(loader):
logging.info(f"====================================")
logging.info(f"====================================")
logging.info(f"now processing mega step {m+1}/{mega_n_steps}")
lengths = np.array(mega_batch['end_time']) - np.array(mega_batch['begin_time'])
sorted_inds = sort_by_audio_len(lengths)
for j in range(len(sorted_inds))[::-1]:
if lengths[sorted_inds[j]] < 0.2 or lengths[sorted_inds[j]] > args.len_cap: # skip samples that are too short (shorter than 0.2s), or too big (bigger than 80s)
skip += 1
del sorted_inds[j]
n_steps = int(np.ceil(len(sorted_inds) / args.batch_size))
for n in tqdm.tqdm(range(n_steps), disable=True):
inds_used = sorted_inds[n*args.batch_size:(n+1)*args.batch_size]
audio_batch = [mega_batch['audio'][id] for id in inds_used]
sr_batch = [mega_batch['sr'][id] for id in inds_used]
segment_id_batch = [mega_batch['segment_id'][id] for id in inds_used]
text_batch = [mega_batch['text'][id] for id in inds_used]
padded_wav = torch.nn.utils.rnn.pad_sequence(audio_batch, batch_first=True).unsqueeze(1) # [B, T] -> [B, 1, T]
all_lens = [lengths[id] for id in inds_used]
with torch.no_grad():
if max(all_lens) > args.max_len and len(all_lens) > 1: # NOTE decrease args.max_len if OOM, or chunk it into more than 2 forward passes
codes = []
inwav = padded_wav.cuda()
codes.append(model.encode(inwav[:len(inwav)//2])[0].cpu())
codes.append(model.encode(inwav[len(inwav)//2:])[0].cpu())
codes = torch.cat(codes, dim=0)
else:
encoded_frames = model.encode(padded_wav.cuda())
# logging.info(f"encoded_frames: {encoded_frames[0].shape}")
codes = encoded_frames[0].cpu()
for i, length in enumerate(all_lens):
save_fn = os.path.join(codes_save_root, segment_id_batch[i]+".txt")
actual_len = round(length * args.model_code_sr) # 320 is downsample rate for this model
cur_code = codes[i].tolist() if type(codes) == list else codes[i, :, :actual_len].tolist()
write_array_to_txt_file(cur_code, save_fn)