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- 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
- # 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 = (
- "<green>{time:YYYY-MM-DD HH:mm:ss.SSS}</green> | "
- "<level>{level: <8}</level> | "
- "<cyan>{name}</cyan>:<cyan>{function}</cyan>:<cyan>{line}</cyan> | "
- "{extra[rank]} - <level>{message}</level>"
- )
- 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)
- def main(
- folder: str,
- num_workers: int,
- config_name: str,
- checkpoint_path: str,
- batch_size: int,
- ):
- 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")
- 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()
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