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inference_save_code.py
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import os
import librosa
import torch
import torch.nn.functional as F
import numpy as np
import soundfile as sf
from glob import glob
from tqdm import tqdm
from os.path import basename, join, exists
from vq.codec_encoder import CodecEncoder
from vq.codec_decoder_vocos import CodecDecoderVocos
from argparse import ArgumentParser
from time import time
from transformers import AutoModel, AutoFeatureExtractor, Wav2Vec2BertModel
import torch.nn as nn
from vq.module import SemanticDecoder, SemanticEncoder
from torch.utils.data import Dataset, DataLoader
from typing import List, Tuple
from collections import OrderedDict
import torchaudio
from torchaudio.transforms import Resample
import pandas as pd
import numpy as np
def pad_audio_batch(batch):
audio_list, feat_list, fname_list, audio_length = zip(*batch)
feat_list = list(feat_list)
max_length_feat = max([feat.shape[1] for feat in feat_list])
max_length = max_length_feat *320
padded_audios = []
for audio in audio_list:
padding = max_length - audio.shape[1]
if padding > 0:
padded_audio = F.pad(audio, (0, padding) , mode='constant', value=0)
else:
padded_audio = audio[:,:max_length]
padded_audios.append(padded_audio)
padded_audios = torch.stack(padded_audios)
padded_feat_list = []
for feat in feat_list:
padding = max_length_feat - feat.shape[1]
padded_feat = F.pad(feat, (0, 0, 0, padding), mode='constant', value=0)
padded_feat_list.append(padded_feat)
padded_feat_list = torch.stack(padded_feat_list)
return torch.tensor(padded_audios),torch.tensor(padded_feat_list), fname_list,audio_length
class WaveDataset(Dataset):
def __init__(
self,
file_list,
sampling_rate,
audio_norm_scale: float = 1.0,
root_dir: str = ""
):
self.file_list = file_list
self.sampling_rate = sampling_rate
self.audio_norm_scale = audio_norm_scale
self.hop_length = 320
self.root_dir = root_dir
self.feature_extractor = AutoFeatureExtractor.from_pretrained("facebook/w2v-bert-2.0")
def __getitem__(self, index):
fname = self.file_list[index]
fname = os.path.join(self.root_dir, fname)
audio, sr = torchaudio.load(fname)
if sr != self.sampling_rate:
audio = Resample(sr, self.sampling_rate)(audio)
if self.audio_norm_scale < 1.0:
audio = audio * self.audio_norm_scale
audio_pad = F.pad(audio, (160, 160))
feat = self.feature_extractor(
audio_pad,
sampling_rate=self.sampling_rate,
return_tensors="pt"
).data['input_features']
return audio,feat, fname, int(audio.shape[1] / self.hop_length)
def __len__(self):
return len(self.file_list)
def save_vq_code(vq_codes: torch.Tensor, wav_paths: List[str], lengths: List[int], output_dir: str ):
for i, wav_path in enumerate(wav_paths):
relative_path = os.path.relpath(wav_path, args.input_dir)
code_path = os.path.join(output_dir, 'vq_codes', relative_path.replace('.flac', '.npy'))
os.makedirs(os.path.dirname(code_path), exist_ok=True)
vq_code = vq_codes[i, 0,:lengths[i]]
np.save(code_path, vq_code.detach().cpu().numpy().astype(np.int32))
if __name__ == '__main__':
parser = ArgumentParser()
parser.add_argument('--local-rank', type=int, default=0, help='Local GPU device ID')
parser.add_argument('--input-dir', type=str, default='/path/to/audio_folder', help='Input directory containing audio files')
parser.add_argument('--flist_file', type=str, default='/path/to/file.txt', help='TSV file containing paths to audio files')
parser.add_argument('--ckpt', type=str, default='/path/to/epoch=4-step=1400000.ckpt', help='Path to the model checkpoint')
parser.add_argument('--output-dir', type=str, default='/path/to/saving_code_folder', help='Output directory for saving audio files')
parser.add_argument('--batch_size', type=int, default=6, help='Batch size for processing')
parser.add_argument('--num_workers', type=int, default=4, help='Number of worker threads for the DataLoader')
device_id = int(os.getenv('LOCAL_RANK', 0))
args = parser.parse_args()
sr = 16000
os.makedirs(args.output_dir, exist_ok=True)
print(f'loading codec checkpoint from {args.ckpt}')
ckpt = torch.load(args.ckpt, map_location='cpu')
ckpt = ckpt['state_dict']
filtered_state_dict_codec = OrderedDict()
filtered_state_dict_semantic_encoder = OrderedDict()
filtered_state_dict_gen = OrderedDict()
filtered_state_dict_fc_post_a = OrderedDict()
filtered_state_dict_fc_prior = OrderedDict()
for key, value in ckpt.items():
if key.startswith('CodecEnc.'):
new_key = key[len('CodecEnc.'):]
filtered_state_dict_codec[new_key] = value
elif key.startswith('generator.'):
new_key = key[len('generator.'):]
filtered_state_dict_gen[new_key] = value
elif key.startswith('fc_post_a.'):
new_key = key[len('fc_post_a.'):]
filtered_state_dict_fc_post_a[new_key] = value
elif key.startswith('SemanticEncoder_module.'):
new_key = key[len('SemanticEncoder_module.'):]
filtered_state_dict_semantic_encoder[new_key] = value
elif key.startswith('fc_prior.'):
new_key = key[len('fc_prior.'):]
filtered_state_dict_fc_prior[new_key] = value
semantic_model = Wav2Vec2BertModel.from_pretrained("facebook/w2v-bert-2.0", output_hidden_states=True)
semantic_model.eval()
SemanticEncoder_module = SemanticEncoder(1024, 1024, 1024)
SemanticEncoder_module.load_state_dict(filtered_state_dict_semantic_encoder)
SemanticEncoder_module.eval()
encoder = CodecEncoder()
encoder.load_state_dict(filtered_state_dict_codec)
encoder.eval()
decoder = CodecDecoderVocos()
decoder.load_state_dict(filtered_state_dict_gen)
decoder.eval()
fc_post_a = nn.Linear(2048, 1024)
fc_post_a.load_state_dict(filtered_state_dict_fc_post_a)
fc_post_a.eval()
fc_prior = nn.Linear(2048, 2048)
fc_prior.load_state_dict(filtered_state_dict_fc_prior)
fc_prior.eval()
device = torch.device(f'cuda:{device_id}' if torch.cuda.is_available() else 'cpu')
semantic_model.to(device)
SemanticEncoder_module.to(device)
encoder.to(device)
decoder.to(device)
fc_post_a.to(device)
fc_prior.to(device)
# assume 8 gpus on your devices, flist_file can obtained using get_tsv.py
df = pd.read_csv(args.flist_file, sep='\t', header=None, names=['filename', 'duration'], skiprows=1)
file_list = df['filename'].tolist()
# with open(args.flist_file, 'r') as f:
# file_list = [line.strip() for line in f if line.strip()]
split_file_lists = np.array_split(file_list, 8) #8 gpus
device_id = device_id
current_file_list = split_file_lists[device_id]
dataset = WaveDataset(file_list=current_file_list, sampling_rate=sr, root_dir=args.input_dir)
dataloader = DataLoader(
dataset,
batch_size=args.batch_size,
shuffle=False,
num_workers=args.num_workers,
pin_memory=True,
collate_fn=pad_audio_batch
)
st = time()
for batch in tqdm(dataloader, desc="processing"):
wavs,feats,wav_paths, lengths = batch
wavs = wavs.to(device)
with torch.no_grad():
vq_emb = encoder(wavs )
vq_emb = vq_emb.transpose(1, 2)
semantic_target = semantic_model(feats[:,0,:,:].to(device))
semantic_target = semantic_target.hidden_states[16]
semantic_target = semantic_target.transpose(1, 2)
semantic_target = SemanticEncoder_module(semantic_target)
vq_emb = torch.cat([semantic_target, vq_emb], dim=1)
vq_emb = fc_prior(vq_emb.transpose(1, 2)).transpose(1, 2)
_, vq_code, _ = decoder(vq_emb, vq=True)
save_vq_code(vq_code, wav_paths, lengths, args.output_dir)
et = time()
print(f'End,time: {(et - st)/60:.2f} mins')