import os
import subprocess as sp
import sys
import time
from datetime import timedelta
from functools import lru_cache
from pathlib import Path
from random import Random
import click
import numpy as np
import torch
import torchaudio
from einops import rearrange
from hydra import compose, initialize
from hydra.utils import instantiate
from lightning import LightningModule
from loguru import logger
from omegaconf import OmegaConf
from fish_speech.models.vqgan.utils import sequence_mask
from fish_speech.utils.file import AUDIO_EXTENSIONS, list_files, load_filelist
# register eval resolver
OmegaConf.register_new_resolver("eval", eval)
# This file is used to convert the audio files to text files using the Whisper model.
# It's mainly used to generate the training data for the VQ model.
RANK = int(os.environ.get("SLURM_PROCID", 0))
WORLD_SIZE = int(os.environ.get("SLURM_NTASKS", 1))
logger_format = (
"{time:YYYY-MM-DD HH:mm:ss.SSS} | "
"{level: <8} | "
"{name}:{function}:{line} | "
"{extra[rank]} - {message}"
)
logger.configure(extra={"rank": f"RANK: {RANK} / {WORLD_SIZE}"})
logger.remove()
logger.add(sys.stderr, format=logger_format)
@lru_cache(maxsize=1)
def get_model(
config_name: str = "vqgan",
checkpoint_path: str = "checkpoints/vqgan/step_000380000.ckpt",
):
with initialize(version_base="1.3", config_path="../../fish_speech/configs"):
cfg = compose(config_name=config_name)
model: LightningModule = instantiate(cfg.model)
state_dict = torch.load(
checkpoint_path,
map_location=model.device,
)
if "state_dict" in state_dict:
state_dict = state_dict["state_dict"]
model.load_state_dict(state_dict, strict=True)
model.eval()
model.cuda()
logger.info(f"Loaded model")
return model
def process_batch(files: list[Path], model) -> float:
wavs = []
audio_lengths = []
max_length = total_time = 0
for file in files:
wav, sr = torchaudio.load(file)
if wav.shape[0] > 1:
wav = wav.mean(dim=0, keepdim=True)
wav = torchaudio.functional.resample(wav.cuda(), sr, model.sampling_rate)[0]
wavs.append(wav)
total_time += len(wav) / model.sampling_rate
max_length = max(max_length, len(wav))
audio_lengths.append(len(wav))
# Pad to max length
for i, wav in enumerate(wavs):
wavs[i] = torch.nn.functional.pad(wav, (0, max_length - len(wav)), "constant")
audios = torch.stack(wavs, dim=0)[:, None]
audio_lengths = torch.tensor(audio_lengths, device=model.device, dtype=torch.long)
# Calculate lengths
with torch.no_grad():
# VQ Encoder
features = gt_mels = model.mel_transform(
audios, sample_rate=model.sampling_rate
)
if model.downsample is not None:
features = model.downsample(features)
feature_lengths = (
audio_lengths
/ model.hop_length
/ (model.downsample.total_strides if model.downsample is not None else 1)
).long()
feature_masks = torch.unsqueeze(
sequence_mask(feature_lengths, features.shape[2]), 1
).to(gt_mels.dtype)
text_features = model.mel_encoder(features, feature_masks)
_, indices, _ = model.vq_encoder(text_features, feature_masks)
if indices.ndim == 4:
# Grouped vq
assert indices.shape[-1] == 1, f"Residual vq is not supported"
indices = indices.squeeze(-1)
elif indices.ndim == 2:
# Single vq
indices = indices.unsqueeze(0)
else:
raise ValueError(f"Invalid indices shape {indices.shape}")
indices = rearrange(indices, "c b t -> b c t")
# Save to disk
outputs = indices.cpu().numpy()
for file, length, feature, audio in zip(files, feature_lengths, outputs, audios):
feature = feature[:, :length]
# (T,)
with open(file.with_suffix(".npy"), "wb") as f:
np.save(f, feature)
return total_time
@click.command()
@click.argument("folder")
@click.option("--num-workers", default=1)
@click.option("--config-name", default="vqgan_pretrain")
@click.option(
"--checkpoint-path",
default="checkpoints/vqgan-v1.pth",
)
@click.option("--batch-size", default=64)
@click.option("--filelist", default=None, type=Path)
def main(
folder: str,
num_workers: int,
config_name: str,
checkpoint_path: str,
batch_size: int,
filelist: Path,
):
if num_workers > 1 and WORLD_SIZE != num_workers:
assert WORLD_SIZE == 1, "You should either use SLURM or this launcher, not both"
logger.info(f"Spawning {num_workers} workers")
visible_devices = os.environ.get("CUDA_VISIBLE_DEVICES", None)
if visible_devices is None:
visible_devices = list(range(torch.cuda.device_count()))
else:
visible_devices = visible_devices.split(",")
processes = []
for i in range(num_workers):
env = os.environ.copy()
env["CUDA_VISIBLE_DEVICES"] = str(visible_devices[i % len(visible_devices)])
env["SLURM_PROCID"] = str(i)
env["SLURM_NTASKS"] = str(num_workers)
processes.append(
sp.Popen(
[sys.executable] + sys.argv.copy(),
env=env,
)
)
for p in processes:
p.wait()
logger.info(f"All workers finished")
return
# This is a worker
logger.info(f"Starting worker")
if filelist:
files = [i[0] for i in load_filelist(filelist)]
else:
files = list_files(folder, AUDIO_EXTENSIONS, recursive=True, sort=True)
Random(42).shuffle(files)
total_files = len(files)
files = files[RANK::WORLD_SIZE]
logger.info(f"Processing {len(files)}/{total_files} files")
# Batch processing
total_time = 0
begin_time = time.time()
processed_files = 0
model = get_model(config_name, checkpoint_path)
for n_batch, idx in enumerate(range(0, len(files), batch_size)):
batch = files[idx : idx + batch_size]
batch_time = process_batch(batch, model)
total_time += batch_time
processed_files += len(batch)
if (n_batch + 1) % 10 == 0:
eta = (
(time.time() - begin_time)
/ processed_files
* (len(files) - processed_files)
)
logger.info(
f"Processed {processed_files} files, {total_time / 3600:.2f} hours of audio, "
+ f"ETA: {timedelta(seconds=round(eta))}s"
)
logger.info(
f"Finished processing {len(files)} files, {total_time / 3600:.2f} hours of audio"
)
if __name__ == "__main__":
main()