FQiao's picture
Upload 70 files
3324de2 verified
raw
history blame contribute delete
20.8 kB
import time
import argparse
import json
import logging
import math
import os
import yaml
# from tqdm import tqdm
import copy
from pathlib import Path
import diffusers
import datasets
import numpy as np
import pandas as pd
import wandb
import transformers
import torch
from accelerate import Accelerator
from accelerate.logging import get_logger
from accelerate.utils import set_seed
from datasets import load_dataset
from torch.utils.data import Dataset, DataLoader
from tqdm.auto import tqdm
from transformers import SchedulerType, get_scheduler
from tangoflux.model import TangoFlux
from datasets import load_dataset, Audio
from tangoflux.utils import Text2AudioDataset, read_wav_file, DPOText2AudioDataset
from diffusers import AutoencoderOobleck
import torchaudio
logger = get_logger(__name__)
def parse_args():
parser = argparse.ArgumentParser(
description="Rectified flow for text to audio generation task."
)
parser.add_argument(
"--num_examples",
type=int,
default=-1,
help="How many examples to use for training and validation.",
)
parser.add_argument(
"--text_column",
type=str,
default="captions",
help="The name of the column in the datasets containing the input texts.",
)
parser.add_argument(
"--audio_column",
type=str,
default="location",
help="The name of the column in the datasets containing the audio paths.",
)
parser.add_argument(
"--adam_beta1",
type=float,
default=0.9,
help="The beta1 parameter for the Adam optimizer.",
)
parser.add_argument(
"--adam_beta2",
type=float,
default=0.95,
help="The beta2 parameter for the Adam optimizer.",
)
parser.add_argument(
"--config",
type=str,
default="tangoflux_config.yaml",
help="Config file defining the model size.",
)
parser.add_argument(
"--weight_decay", type=float, default=1e-8, help="Weight decay to use."
)
parser.add_argument(
"--max_train_steps",
type=int,
default=None,
help="Total number of training steps to perform. If provided, overrides num_train_epochs.",
)
parser.add_argument(
"--lr_scheduler_type",
type=SchedulerType,
default="linear",
help="The scheduler type to use.",
choices=[
"linear",
"cosine",
"cosine_with_restarts",
"polynomial",
"constant",
"constant_with_warmup",
],
)
parser.add_argument(
"--adam_epsilon",
type=float,
default=1e-08,
help="Epsilon value for the Adam optimizer",
)
parser.add_argument(
"--adam_weight_decay",
type=float,
default=1e-2,
help="Epsilon value for the Adam optimizer",
)
parser.add_argument(
"--seed", type=int, default=None, help="A seed for reproducible training."
)
parser.add_argument(
"--checkpointing_steps",
type=str,
default="best",
help="Whether the various states should be saved at the end of every 'epoch' or 'best' whenever validation loss decreases.",
)
parser.add_argument(
"--save_every",
type=int,
default=5,
help="Save model after every how many epochs when checkpointing_steps is set to best.",
)
parser.add_argument(
"--load_from_checkpoint",
type=str,
default=None,
help="Whether to continue training from a model weight",
)
args = parser.parse_args()
# Sanity checks
# if args.train_file is None and args.validation_file is None:
# raise ValueError("Need a training/validation file.")
# else:
# if args.train_file is not None:
# extension = args.train_file.split(".")[-1]
# assert extension in ["csv", "json"], "`train_file` should be a csv or a json file."
# if args.validation_file is not None:
# extension = args.validation_file.split(".")[-1]
# assert extension in ["csv", "json"], "`validation_file` should be a csv or a json file."
return args
def main():
args = parse_args()
accelerator_log_kwargs = {}
def load_config(config_path):
with open(config_path, "r") as file:
return yaml.safe_load(file)
config = load_config(args.config)
learning_rate = float(config["training"]["learning_rate"])
num_train_epochs = int(config["training"]["num_train_epochs"])
num_warmup_steps = int(config["training"]["num_warmup_steps"])
per_device_batch_size = int(config["training"]["per_device_batch_size"])
gradient_accumulation_steps = int(config["training"]["gradient_accumulation_steps"])
output_dir = config["paths"]["output_dir"]
accelerator = Accelerator(
gradient_accumulation_steps=gradient_accumulation_steps,
**accelerator_log_kwargs,
)
# Make one log on every process with the configuration for debugging.
logging.basicConfig(
format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
datefmt="%m/%d/%Y %H:%M:%S",
level=logging.INFO,
)
logger.info(accelerator.state, main_process_only=False)
datasets.utils.logging.set_verbosity_error()
diffusers.utils.logging.set_verbosity_error()
transformers.utils.logging.set_verbosity_error()
# If passed along, set the training seed now.
if args.seed is not None:
set_seed(args.seed)
# Handle output directory creation and wandb tracking
if accelerator.is_main_process:
if output_dir is None or output_dir == "":
output_dir = "saved/" + str(int(time.time()))
if not os.path.exists("saved"):
os.makedirs("saved")
os.makedirs(output_dir, exist_ok=True)
elif output_dir is not None:
os.makedirs(output_dir, exist_ok=True)
os.makedirs("{}/{}".format(output_dir, "outputs"), exist_ok=True)
with open("{}/summary.jsonl".format(output_dir), "a") as f:
f.write(json.dumps(dict(vars(args))) + "\n\n")
accelerator.project_configuration.automatic_checkpoint_naming = False
wandb.init(
project="Text to Audio Flow matching DPO",
settings=wandb.Settings(_disable_stats=True),
)
accelerator.wait_for_everyone()
# Get the datasets
data_files = {}
# if args.train_file is not None:
if config["paths"]["train_file"] != "":
data_files["train"] = config["paths"]["train_file"]
# if args.validation_file is not None:
if config["paths"]["val_file"] != "":
data_files["validation"] = config["paths"]["val_file"]
if config["paths"]["test_file"] != "":
data_files["test"] = config["paths"]["test_file"]
else:
data_files["test"] = config["paths"]["val_file"]
extension = "json"
train_dataset = load_dataset(extension, data_files=data_files["train"])
data_files.pop("train")
raw_datasets = load_dataset(extension, data_files=data_files)
text_column, audio_column = args.text_column, args.audio_column
model = TangoFlux(config=config["model"], initialize_reference_model=True)
vae = AutoencoderOobleck.from_pretrained(
"stabilityai/stable-audio-open-1.0", subfolder="vae"
)
## Freeze vae
for param in vae.parameters():
vae.requires_grad = False
vae.eval()
## Freeze text encoder param
for param in model.text_encoder.parameters():
param.requires_grad = False
model.text_encoder.eval()
prefix = ""
with accelerator.main_process_first():
train_dataset = DPOText2AudioDataset(
train_dataset["train"],
prefix,
text_column,
"chosen",
"reject",
"duration",
args.num_examples,
)
eval_dataset = Text2AudioDataset(
raw_datasets["validation"],
prefix,
text_column,
audio_column,
"duration",
args.num_examples,
)
test_dataset = Text2AudioDataset(
raw_datasets["test"],
prefix,
text_column,
audio_column,
"duration",
args.num_examples,
)
accelerator.print(
"Num instances in train: {}, validation: {}, test: {}".format(
train_dataset.get_num_instances(),
eval_dataset.get_num_instances(),
test_dataset.get_num_instances(),
)
)
train_dataloader = DataLoader(
train_dataset,
shuffle=True,
batch_size=config["training"]["per_device_batch_size"],
collate_fn=train_dataset.collate_fn,
)
eval_dataloader = DataLoader(
eval_dataset,
shuffle=True,
batch_size=config["training"]["per_device_batch_size"],
collate_fn=eval_dataset.collate_fn,
)
test_dataloader = DataLoader(
test_dataset,
shuffle=False,
batch_size=config["training"]["per_device_batch_size"],
collate_fn=test_dataset.collate_fn,
)
# Optimizer
optimizer_parameters = list(model.transformer.parameters()) + list(
model.fc.parameters()
)
num_trainable_parameters = sum(
p.numel() for p in model.parameters() if p.requires_grad
)
accelerator.print("Num trainable parameters: {}".format(num_trainable_parameters))
if args.load_from_checkpoint:
from safetensors.torch import load_file
w1 = load_file(args.load_from_checkpoint)
model.load_state_dict(w1, strict=False)
logger.info("Weights loaded from{}".format(args.load_from_checkpoint))
import copy
model.ref_transformer = copy.deepcopy(model.transformer)
model.ref_transformer.requires_grad_ = False
model.ref_transformer.eval()
for param in model.ref_transformer.parameters():
param.requires_grad = False
optimizer = torch.optim.AdamW(
optimizer_parameters,
lr=learning_rate,
betas=(args.adam_beta1, args.adam_beta2),
weight_decay=args.adam_weight_decay,
eps=args.adam_epsilon,
)
# Scheduler and math around the number of training steps.
overrode_max_train_steps = False
num_update_steps_per_epoch = math.ceil(
len(train_dataloader) / gradient_accumulation_steps
)
if args.max_train_steps is None:
args.max_train_steps = num_train_epochs * num_update_steps_per_epoch
overrode_max_train_steps = True
lr_scheduler = get_scheduler(
name=args.lr_scheduler_type,
optimizer=optimizer,
num_warmup_steps=num_warmup_steps
* gradient_accumulation_steps
* accelerator.num_processes,
num_training_steps=args.max_train_steps * gradient_accumulation_steps,
)
# Prepare everything with our `accelerator`.
vae, model, optimizer, lr_scheduler = accelerator.prepare(
vae, model, optimizer, lr_scheduler
)
train_dataloader, eval_dataloader, test_dataloader = accelerator.prepare(
train_dataloader, eval_dataloader, test_dataloader
)
# We need to recalculate our total training steps as the size of the training dataloader may have changed.
num_update_steps_per_epoch = math.ceil(
len(train_dataloader) / gradient_accumulation_steps
)
if overrode_max_train_steps:
args.max_train_steps = num_train_epochs * num_update_steps_per_epoch
# Afterwards we recalculate our number of training epochs
num_train_epochs = math.ceil(args.max_train_steps / num_update_steps_per_epoch)
# Figure out how many steps we should save the Accelerator states
checkpointing_steps = args.checkpointing_steps
if checkpointing_steps is not None and checkpointing_steps.isdigit():
checkpointing_steps = int(checkpointing_steps)
# Train!
total_batch_size = (
per_device_batch_size * accelerator.num_processes * gradient_accumulation_steps
)
logger.info("***** Running training *****")
logger.info(f" Num examples = {len(train_dataset)}")
logger.info(f" Num Epochs = {num_train_epochs}")
logger.info(f" Instantaneous batch size per device = {per_device_batch_size}")
logger.info(
f" Total train batch size (w. parallel, distributed & accumulation) = {total_batch_size}"
)
logger.info(f" Gradient Accumulation steps = {gradient_accumulation_steps}")
logger.info(f" Total optimization steps = {args.max_train_steps}")
# Only show the progress bar once on each machine.
progress_bar = tqdm(
range(args.max_train_steps), disable=not accelerator.is_local_main_process
)
completed_steps = 0
starting_epoch = 0
# Potentially load in the weights and states from a previous save
resume_from_checkpoint = config["paths"]["resume_from_checkpoint"]
if resume_from_checkpoint != "":
accelerator.load_state(resume_from_checkpoint)
accelerator.print(f"Resumed from local checkpoint: {resume_from_checkpoint}")
# Duration of the audio clips in seconds
best_loss = np.inf
length = config["training"]["max_audio_duration"]
for epoch in range(starting_epoch, num_train_epochs):
model.train()
total_loss, total_val_loss = 0, 0
for step, batch in enumerate(train_dataloader):
optimizer.zero_grad()
with accelerator.accumulate(model):
optimizer.zero_grad()
device = accelerator.device
text, audio_w, audio_l, duration, _ = batch
with torch.no_grad():
audio_list_w = []
audio_list_l = []
for audio_path in audio_w:
wav = read_wav_file(
audio_path, length
) ## Only read the first 30 seconds of audio
if (
wav.shape[0] == 1
): ## If this audio is mono, we repeat the channel so it become "fake stereo"
wav = wav.repeat(2, 1)
audio_list_w.append(wav)
for audio_path in audio_l:
wav = read_wav_file(
audio_path, length
) ## Only read the first 30 seconds of audio
if (
wav.shape[0] == 1
): ## If this audio is mono, we repeat the channel so it become "fake stereo"
wav = wav.repeat(2, 1)
audio_list_l.append(wav)
audio_input_w = torch.stack(audio_list_w, dim=0).to(device)
audio_input_l = torch.stack(audio_list_l, dim=0).to(device)
# audio_input_ = audio_input.to(device)
unwrapped_vae = accelerator.unwrap_model(vae)
duration = torch.tensor(duration, device=device)
duration = torch.clamp(
duration, max=length
) ## max duration is 30 sec
audio_latent_w = unwrapped_vae.encode(
audio_input_w
).latent_dist.sample()
audio_latent_l = unwrapped_vae.encode(
audio_input_l
).latent_dist.sample()
audio_latent = torch.cat((audio_latent_w, audio_latent_l), dim=0)
audio_latent = audio_latent.transpose(
1, 2
) ## Tranpose to (bsz, seq_len, channel)
loss, raw_model_loss, raw_ref_loss, implicit_acc = model(
audio_latent, text, duration=duration, sft=False
)
total_loss += loss.detach().float()
accelerator.backward(loss)
optimizer.step()
lr_scheduler.step()
# if accelerator.sync_gradients:
if accelerator.sync_gradients:
# accelerator.clip_grad_value_(model.parameters(),1.0)
progress_bar.update(1)
completed_steps += 1
if completed_steps % 10 == 0 and accelerator.is_main_process:
total_norm = 0.0
for p in model.parameters():
if p.grad is not None:
param_norm = p.grad.data.norm(2)
total_norm += param_norm.item() ** 2
total_norm = total_norm**0.5
logger.info(
f"Step {completed_steps}, Loss: {loss.item()}, Grad Norm: {total_norm}"
)
lr = lr_scheduler.get_last_lr()[0]
result = {
"train_loss": loss.item(),
"grad_norm": total_norm,
"learning_rate": lr,
"raw_model_loss": raw_model_loss,
"raw_ref_loss": raw_ref_loss,
"implicit_acc": implicit_acc,
}
# result["val_loss"] = round(total_val_loss.item()/len(eval_dataloader), 4)
wandb.log(result, step=completed_steps)
# Checks if the accelerator has performed an optimization step behind the scenes
if isinstance(checkpointing_steps, int):
if completed_steps % checkpointing_steps == 0:
output_dir = f"step_{completed_steps }"
if output_dir is not None:
output_dir = os.path.join(output_dir, output_dir)
accelerator.save_state(output_dir)
if completed_steps >= args.max_train_steps:
break
model.eval()
eval_progress_bar = tqdm(
range(len(eval_dataloader)), disable=not accelerator.is_local_main_process
)
for step, batch in enumerate(eval_dataloader):
with accelerator.accumulate(model) and torch.no_grad():
device = model.device
text, audios, duration, _ = batch
audio_list = []
for audio_path in audios:
wav = read_wav_file(
audio_path, length
) ## Only read the first 30 seconds of audio
if (
wav.shape[0] == 1
): ## If this audio is mono, we repeat the channel so it become "fake stereo"
wav = wav.repeat(2, 1)
audio_list.append(wav)
audio_input = torch.stack(audio_list, dim=0)
audio_input = audio_input.to(device)
duration = torch.tensor(duration, device=device)
unwrapped_vae = accelerator.unwrap_model(vae)
audio_latent = unwrapped_vae.encode(audio_input).latent_dist.sample()
audio_latent = audio_latent.transpose(
1, 2
) ## Tranpose to (bsz, seq_len, channel)
val_loss, _, _, _ = model(
audio_latent, text, duration=duration, sft=True
)
total_val_loss += val_loss.detach().float()
eval_progress_bar.update(1)
if accelerator.is_main_process:
result = {}
result["epoch"] = float(epoch + 1)
result["epoch/train_loss"] = round(
total_loss.item() / len(train_dataloader), 4
)
result["epoch/val_loss"] = round(
total_val_loss.item() / len(eval_dataloader), 4
)
wandb.log(result, step=completed_steps)
with open("{}/summary.jsonl".format(output_dir), "a") as f:
f.write(json.dumps(result) + "\n\n")
logger.info(result)
save_checkpoint = True
accelerator.wait_for_everyone()
if accelerator.is_main_process and args.checkpointing_steps == "best":
if save_checkpoint:
accelerator.save_state("{}/{}".format(output_dir, "best"))
if (epoch + 1) % args.save_every == 0:
accelerator.save_state(
"{}/{}".format(output_dir, "epoch_" + str(epoch + 1))
)
if accelerator.is_main_process and args.checkpointing_steps == "epoch":
accelerator.save_state(
"{}/{}".format(output_dir, "epoch_" + str(epoch + 1))
)
if __name__ == "__main__":
main()