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Sam Khoze
2024-06-18 13:21:08 -07:00
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commit dc8b8bca5a
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from typing import List
from coqpit import Coqpit
from torch.utils.data import Dataset
from TTS.utils.audio import AudioProcessor
from TTS.vocoder.datasets.gan_dataset import GANDataset
from TTS.vocoder.datasets.preprocess import load_wav_data, load_wav_feat_data
from TTS.vocoder.datasets.wavegrad_dataset import WaveGradDataset
from TTS.vocoder.datasets.wavernn_dataset import WaveRNNDataset
def setup_dataset(config: Coqpit, ap: AudioProcessor, is_eval: bool, data_items: List, verbose: bool) -> Dataset:
if config.model.lower() in "gan":
dataset = GANDataset(
ap=ap,
items=data_items,
seq_len=config.seq_len,
hop_len=ap.hop_length,
pad_short=config.pad_short,
conv_pad=config.conv_pad,
return_pairs=config.diff_samples_for_G_and_D if "diff_samples_for_G_and_D" in config else False,
is_training=not is_eval,
return_segments=not is_eval,
use_noise_augment=config.use_noise_augment,
use_cache=config.use_cache,
verbose=verbose,
)
dataset.shuffle_mapping()
elif config.model.lower() == "wavegrad":
dataset = WaveGradDataset(
ap=ap,
items=data_items,
seq_len=config.seq_len,
hop_len=ap.hop_length,
pad_short=config.pad_short,
conv_pad=config.conv_pad,
is_training=not is_eval,
return_segments=True,
use_noise_augment=False,
use_cache=config.use_cache,
verbose=verbose,
)
elif config.model.lower() == "wavernn":
dataset = WaveRNNDataset(
ap=ap,
items=data_items,
seq_len=config.seq_len,
hop_len=ap.hop_length,
pad=config.model_params.pad,
mode=config.model_params.mode,
mulaw=config.model_params.mulaw,
is_training=not is_eval,
verbose=verbose,
)
else:
raise ValueError(f" [!] Dataset for model {config.model.lower()} cannot be found.")
return dataset
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import glob
import os
import random
from multiprocessing import Manager
import numpy as np
import torch
from torch.utils.data import Dataset
class GANDataset(Dataset):
"""
GAN Dataset searchs for all the wav files under root path
and converts them to acoustic features on the fly and returns
random segments of (audio, feature) couples.
"""
def __init__(
self,
ap,
items,
seq_len,
hop_len,
pad_short,
conv_pad=2,
return_pairs=False,
is_training=True,
return_segments=True,
use_noise_augment=False,
use_cache=False,
verbose=False,
):
super().__init__()
self.ap = ap
self.item_list = items
self.compute_feat = not isinstance(items[0], (tuple, list))
self.seq_len = seq_len
self.hop_len = hop_len
self.pad_short = pad_short
self.conv_pad = conv_pad
self.return_pairs = return_pairs
self.is_training = is_training
self.return_segments = return_segments
self.use_cache = use_cache
self.use_noise_augment = use_noise_augment
self.verbose = verbose
assert seq_len % hop_len == 0, " [!] seq_len has to be a multiple of hop_len."
self.feat_frame_len = seq_len // hop_len + (2 * conv_pad)
# map G and D instances
self.G_to_D_mappings = list(range(len(self.item_list)))
self.shuffle_mapping()
# cache acoustic features
if use_cache:
self.create_feature_cache()
def create_feature_cache(self):
self.manager = Manager()
self.cache = self.manager.list()
self.cache += [None for _ in range(len(self.item_list))]
@staticmethod
def find_wav_files(path):
return glob.glob(os.path.join(path, "**", "*.wav"), recursive=True)
def __len__(self):
return len(self.item_list)
def __getitem__(self, idx):
"""Return different items for Generator and Discriminator and
cache acoustic features"""
# set the seed differently for each worker
if torch.utils.data.get_worker_info():
random.seed(torch.utils.data.get_worker_info().seed)
if self.return_segments:
item1 = self.load_item(idx)
if self.return_pairs:
idx2 = self.G_to_D_mappings[idx]
item2 = self.load_item(idx2)
return item1, item2
return item1
item1 = self.load_item(idx)
return item1
def _pad_short_samples(self, audio, mel=None):
"""Pad samples shorter than the output sequence length"""
if len(audio) < self.seq_len:
audio = np.pad(audio, (0, self.seq_len - len(audio)), mode="constant", constant_values=0.0)
if mel is not None and mel.shape[1] < self.feat_frame_len:
pad_value = self.ap.melspectrogram(np.zeros([self.ap.win_length]))[:, 0]
mel = np.pad(
mel,
([0, 0], [0, self.feat_frame_len - mel.shape[1]]),
mode="constant",
constant_values=pad_value.mean(),
)
return audio, mel
def shuffle_mapping(self):
random.shuffle(self.G_to_D_mappings)
def load_item(self, idx):
"""load (audio, feat) couple"""
if self.compute_feat:
# compute features from wav
wavpath = self.item_list[idx]
# print(wavpath)
if self.use_cache and self.cache[idx] is not None:
audio, mel = self.cache[idx]
else:
audio = self.ap.load_wav(wavpath)
mel = self.ap.melspectrogram(audio)
audio, mel = self._pad_short_samples(audio, mel)
else:
# load precomputed features
wavpath, feat_path = self.item_list[idx]
if self.use_cache and self.cache[idx] is not None:
audio, mel = self.cache[idx]
else:
audio = self.ap.load_wav(wavpath)
mel = np.load(feat_path)
audio, mel = self._pad_short_samples(audio, mel)
# correct the audio length wrt padding applied in stft
audio = np.pad(audio, (0, self.hop_len), mode="edge")
audio = audio[: mel.shape[-1] * self.hop_len]
assert (
mel.shape[-1] * self.hop_len == audio.shape[-1]
), f" [!] {mel.shape[-1] * self.hop_len} vs {audio.shape[-1]}"
audio = torch.from_numpy(audio).float().unsqueeze(0)
mel = torch.from_numpy(mel).float().squeeze(0)
if self.return_segments:
max_mel_start = mel.shape[1] - self.feat_frame_len
mel_start = random.randint(0, max_mel_start)
mel_end = mel_start + self.feat_frame_len
mel = mel[:, mel_start:mel_end]
audio_start = mel_start * self.hop_len
audio = audio[:, audio_start : audio_start + self.seq_len]
if self.use_noise_augment and self.is_training and self.return_segments:
audio = audio + (1 / 32768) * torch.randn_like(audio)
return (mel, audio)
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import glob
import os
from pathlib import Path
import numpy as np
from coqpit import Coqpit
from tqdm import tqdm
from TTS.utils.audio import AudioProcessor
from TTS.utils.audio.numpy_transforms import mulaw_encode, quantize
def preprocess_wav_files(out_path: str, config: Coqpit, ap: AudioProcessor):
"""Process wav and compute mel and quantized wave signal.
It is mainly used by WaveRNN dataloader.
Args:
out_path (str): Parent folder path to save the files.
config (Coqpit): Model config.
ap (AudioProcessor): Audio processor.
"""
os.makedirs(os.path.join(out_path, "quant"), exist_ok=True)
os.makedirs(os.path.join(out_path, "mel"), exist_ok=True)
wav_files = find_wav_files(config.data_path)
for path in tqdm(wav_files):
wav_name = Path(path).stem
quant_path = os.path.join(out_path, "quant", wav_name + ".npy")
mel_path = os.path.join(out_path, "mel", wav_name + ".npy")
y = ap.load_wav(path)
mel = ap.melspectrogram(y)
np.save(mel_path, mel)
if isinstance(config.mode, int):
quant = (
mulaw_encode(wav=y, mulaw_qc=config.mode)
if config.model_args.mulaw
else quantize(x=y, quantize_bits=config.mode)
)
np.save(quant_path, quant)
def find_wav_files(data_path, file_ext="wav"):
wav_paths = glob.glob(os.path.join(data_path, "**", f"*.{file_ext}"), recursive=True)
return wav_paths
def find_feat_files(data_path):
feat_paths = glob.glob(os.path.join(data_path, "**", "*.npy"), recursive=True)
return feat_paths
def load_wav_data(data_path, eval_split_size, file_ext="wav"):
wav_paths = find_wav_files(data_path, file_ext=file_ext)
assert len(wav_paths) > 0, f" [!] {data_path} is empty."
np.random.seed(0)
np.random.shuffle(wav_paths)
return wav_paths[:eval_split_size], wav_paths[eval_split_size:]
def load_wav_feat_data(data_path, feat_path, eval_split_size):
wav_paths = find_wav_files(data_path)
feat_paths = find_feat_files(feat_path)
wav_paths.sort(key=lambda x: Path(x).stem)
feat_paths.sort(key=lambda x: Path(x).stem)
assert len(wav_paths) == len(feat_paths), f" [!] {len(wav_paths)} vs {feat_paths}"
for wav, feat in zip(wav_paths, feat_paths):
wav_name = Path(wav).stem
feat_name = Path(feat).stem
assert wav_name == feat_name
items = list(zip(wav_paths, feat_paths))
np.random.seed(0)
np.random.shuffle(items)
return items[:eval_split_size], items[eval_split_size:]
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import glob
import os
import random
from multiprocessing import Manager
from typing import List, Tuple
import numpy as np
import torch
from torch.utils.data import Dataset
class WaveGradDataset(Dataset):
"""
WaveGrad Dataset searchs for all the wav files under root path
and converts them to acoustic features on the fly and returns
random segments of (audio, feature) couples.
"""
def __init__(
self,
ap,
items,
seq_len,
hop_len,
pad_short,
conv_pad=2,
is_training=True,
return_segments=True,
use_noise_augment=False,
use_cache=False,
verbose=False,
):
super().__init__()
self.ap = ap
self.item_list = items
self.seq_len = seq_len if return_segments else None
self.hop_len = hop_len
self.pad_short = pad_short
self.conv_pad = conv_pad
self.is_training = is_training
self.return_segments = return_segments
self.use_cache = use_cache
self.use_noise_augment = use_noise_augment
self.verbose = verbose
if return_segments:
assert seq_len % hop_len == 0, " [!] seq_len has to be a multiple of hop_len."
self.feat_frame_len = seq_len // hop_len + (2 * conv_pad)
# cache acoustic features
if use_cache:
self.create_feature_cache()
def create_feature_cache(self):
self.manager = Manager()
self.cache = self.manager.list()
self.cache += [None for _ in range(len(self.item_list))]
@staticmethod
def find_wav_files(path):
return glob.glob(os.path.join(path, "**", "*.wav"), recursive=True)
def __len__(self):
return len(self.item_list)
def __getitem__(self, idx):
item = self.load_item(idx)
return item
def load_test_samples(self, num_samples: int) -> List[Tuple]:
"""Return test samples.
Args:
num_samples (int): Number of samples to return.
Returns:
List[Tuple]: melspectorgram and audio.
Shapes:
- melspectrogram (Tensor): :math:`[C, T]`
- audio (Tensor): :math:`[T_audio]`
"""
samples = []
return_segments = self.return_segments
self.return_segments = False
for idx in range(num_samples):
mel, audio = self.load_item(idx)
samples.append([mel, audio])
self.return_segments = return_segments
return samples
def load_item(self, idx):
"""load (audio, feat) couple"""
# compute features from wav
wavpath = self.item_list[idx]
if self.use_cache and self.cache[idx] is not None:
audio = self.cache[idx]
else:
audio = self.ap.load_wav(wavpath)
if self.return_segments:
# correct audio length wrt segment length
if audio.shape[-1] < self.seq_len + self.pad_short:
audio = np.pad(
audio, (0, self.seq_len + self.pad_short - len(audio)), mode="constant", constant_values=0.0
)
assert (
audio.shape[-1] >= self.seq_len + self.pad_short
), f"{audio.shape[-1]} vs {self.seq_len + self.pad_short}"
# correct the audio length wrt hop length
p = (audio.shape[-1] // self.hop_len + 1) * self.hop_len - audio.shape[-1]
audio = np.pad(audio, (0, p), mode="constant", constant_values=0.0)
if self.use_cache:
self.cache[idx] = audio
if self.return_segments:
max_start = len(audio) - self.seq_len
start = random.randint(0, max_start)
end = start + self.seq_len
audio = audio[start:end]
if self.use_noise_augment and self.is_training and self.return_segments:
audio = audio + (1 / 32768) * torch.randn_like(audio)
mel = self.ap.melspectrogram(audio)
mel = mel[..., :-1] # ignore the padding
audio = torch.from_numpy(audio).float()
mel = torch.from_numpy(mel).float().squeeze(0)
return (mel, audio)
@staticmethod
def collate_full_clips(batch):
"""This is used in tune_wavegrad.py.
It pads sequences to the max length."""
max_mel_length = max([b[0].shape[1] for b in batch]) if len(batch) > 1 else batch[0][0].shape[1]
max_audio_length = max([b[1].shape[0] for b in batch]) if len(batch) > 1 else batch[0][1].shape[0]
mels = torch.zeros([len(batch), batch[0][0].shape[0], max_mel_length])
audios = torch.zeros([len(batch), max_audio_length])
for idx, b in enumerate(batch):
mel = b[0]
audio = b[1]
mels[idx, :, : mel.shape[1]] = mel
audios[idx, : audio.shape[0]] = audio
return mels, audios
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import numpy as np
import torch
from torch.utils.data import Dataset
from TTS.utils.audio.numpy_transforms import mulaw_encode, quantize
class WaveRNNDataset(Dataset):
"""
WaveRNN Dataset searchs for all the wav files under root path
and converts them to acoustic features on the fly.
"""
def __init__(
self, ap, items, seq_len, hop_len, pad, mode, mulaw, is_training=True, verbose=False, return_segments=True
):
super().__init__()
self.ap = ap
self.compute_feat = not isinstance(items[0], (tuple, list))
self.item_list = items
self.seq_len = seq_len
self.hop_len = hop_len
self.mel_len = seq_len // hop_len
self.pad = pad
self.mode = mode
self.mulaw = mulaw
self.is_training = is_training
self.verbose = verbose
self.return_segments = return_segments
assert self.seq_len % self.hop_len == 0
def __len__(self):
return len(self.item_list)
def __getitem__(self, index):
item = self.load_item(index)
return item
def load_test_samples(self, num_samples):
samples = []
return_segments = self.return_segments
self.return_segments = False
for idx in range(num_samples):
mel, audio, _ = self.load_item(idx)
samples.append([mel, audio])
self.return_segments = return_segments
return samples
def load_item(self, index):
"""
load (audio, feat) couple if feature_path is set
else compute it on the fly
"""
if self.compute_feat:
wavpath = self.item_list[index]
audio = self.ap.load_wav(wavpath)
if self.return_segments:
min_audio_len = 2 * self.seq_len + (2 * self.pad * self.hop_len)
else:
min_audio_len = audio.shape[0] + (2 * self.pad * self.hop_len)
if audio.shape[0] < min_audio_len:
print(" [!] Instance is too short! : {}".format(wavpath))
audio = np.pad(audio, [0, min_audio_len - audio.shape[0] + self.hop_len])
mel = self.ap.melspectrogram(audio)
if self.mode in ["gauss", "mold"]:
x_input = audio
elif isinstance(self.mode, int):
x_input = (
mulaw_encode(wav=audio, mulaw_qc=self.mode)
if self.mulaw
else quantize(x=audio, quantize_bits=self.mode)
)
else:
raise RuntimeError("Unknown dataset mode - ", self.mode)
else:
wavpath, feat_path = self.item_list[index]
mel = np.load(feat_path.replace("/quant/", "/mel/"))
if mel.shape[-1] < self.mel_len + 2 * self.pad:
print(" [!] Instance is too short! : {}".format(wavpath))
self.item_list[index] = self.item_list[index + 1]
feat_path = self.item_list[index]
mel = np.load(feat_path.replace("/quant/", "/mel/"))
if self.mode in ["gauss", "mold"]:
x_input = self.ap.load_wav(wavpath)
elif isinstance(self.mode, int):
x_input = np.load(feat_path.replace("/mel/", "/quant/"))
else:
raise RuntimeError("Unknown dataset mode - ", self.mode)
return mel, x_input, wavpath
def collate(self, batch):
mel_win = self.seq_len // self.hop_len + 2 * self.pad
max_offsets = [x[0].shape[-1] - (mel_win + 2 * self.pad) for x in batch]
mel_offsets = [np.random.randint(0, offset) for offset in max_offsets]
sig_offsets = [(offset + self.pad) * self.hop_len for offset in mel_offsets]
mels = [x[0][:, mel_offsets[i] : mel_offsets[i] + mel_win] for i, x in enumerate(batch)]
coarse = [x[1][sig_offsets[i] : sig_offsets[i] + self.seq_len + 1] for i, x in enumerate(batch)]
mels = np.stack(mels).astype(np.float32)
if self.mode in ["gauss", "mold"]:
coarse = np.stack(coarse).astype(np.float32)
coarse = torch.FloatTensor(coarse)
x_input = coarse[:, : self.seq_len]
elif isinstance(self.mode, int):
coarse = np.stack(coarse).astype(np.int64)
coarse = torch.LongTensor(coarse)
x_input = 2 * coarse[:, : self.seq_len].float() / (2**self.mode - 1.0) - 1.0
y_coarse = coarse[:, 1:]
mels = torch.FloatTensor(mels)
return x_input, mels, y_coarse