SSR-Speech / data /encode.py
OpenSound's picture
11
f5b4ff2
raw
history blame
8.35 kB
# @ [email protected]
import argparse
def parse_args():
parser = argparse.ArgumentParser(description="encode the librilight dataset using encodec model")
parser.add_argument("--audiopath", type=str, default=None)
parser.add_argument('--save_dir', type=str, default=None)
parser.add_argument('--save_tag', type=str, default='encodec_16khz_4codebooks')
parser.add_argument('--dataset_name', type=str, default=None)
parser.add_argument('--encodec_model_path', type=str, default=None)
parser.add_argument('--n_workers', type=int, default=4, help="Number of parallel worker processes")
parser.add_argument('--mega_batch_size', type=int, default=120, help="Number of samples in each mega batch for multiprocess dataloading")
parser.add_argument('--batch_size', type=int, default=32, 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=20.0, help='will drop audios that are longer than this number')
parser.add_argument('--start', type=int, default=0, help='start index for parallel processing')
parser.add_argument('--end', type=int, default=500000, help='end index for parallel processing')
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
os.environ["USER"] = "root"
import numpy as np
import torch
import tqdm
import time
import torchaudio
from datasets import load_dataset, DownloadConfig
import pandas as pd
from tokenizer import TextTokenizer, tokenize_text
import torchaudio.transforms as transforms
# get the path encodec_16khz_4codebooks
codes_save_root = os.path.join(args.save_dir, args.dataset_name, args.save_tag)
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 WMCompressionSolver
model = WMCompressionSolver.model_from_checkpoint(args.encodec_model_path)
model = model.cuda()
model = model.eval()
class mydataset(torch.utils.data.Dataset):
def __init__(self, args, split):
super().__init__()
import glob
self.data = glob.glob(os.path.join(args.audiopath, "*.wav"))
self.data = self.data[args.start:args.end]
def checkout(self, data):
out = []
for ind in range(len(data)):
segment_id = data[ind].split('/')[-1].split(".wav")[0]
save_fn = os.path.join(codes_save_root, segment_id+".txt")
if not os.path.exists(save_fn):
out.append(data[ind])
return out
def __len__(self):
return len(self.data)
def __getitem__(self, ind):
segment_id = self.data[ind].split('/')[-1].split(".wav")[0]
if os.path.exists(self.data[ind]):
audio, sr = torchaudio.load(self.data[ind])
else:
audio, sr = torchaudio.load(self.data[ind].replace('/apdcephfs_cq2', '/apdcephfs_cq2_1297902'))
if sr != 16000:
resampler = transforms.Resample(orig_freq=sr, new_freq=16000)
audio = resampler(audio)
duration = audio.shape[1] / sr
return segment_id, audio.squeeze(), sr, duration
def collate(self, batch):
res = {'segment_id': [], "audio": [], "sr": [], "duration":[]}
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['duration'].append(item[3])
return res
## encodec codes extraction
logging.info("encodec encoding...")
train_dataset = mydataset(args, 'train')
print(len(train_dataset))
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)
splits = ['train']
loaders = [train_loader]
for split, loader in zip(splits, loaders):
skip = 0
logging.info(f"now processing split {split}...")
for m, mega_batch in enumerate(loader):
logging.info(f"====================================")
logging.info(f"====================================")
lengths = np.array(mega_batch['duration'])
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]
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")
if not os.path.exists(save_fn):
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)