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# ==============================================================================
# Training Script for Duration Predictor with Context
#
# This script trains ONLY the Duration Predictor:
# - Predicts phoneme durations based on text and context
# - Uses PredictorDurationSpeakerContext
#
# Validation performs full OOD inference:
# 1. Predict duration (using this model)
# 2. Predict flow latents (using frozen FSQFlowPredictorConvNeXt)
# 3. Decode audio (using frozen Codec)
# ==============================================================================
import os
import os.path as osp
import shutil
import sys
import time
import yaml
import warnings
import logging
import math
import numpy as np
from typing import Optional
import click
import torch
import torch.nn as nn
import torch.nn.functional as F
from torch.utils.tensorboard import SummaryWriter
# Fix for "Too many open files" error
torch.multiprocessing.set_sharing_strategy('file_system')
from accelerate import Accelerator, DistributedDataParallelKwargs
from accelerate.logging import get_logger
from fsq_flow_dataset import build_fsq_flow_dataloader
from utils import get_data_path_list
# Import our predictors
from Modules.predictor_fsq_flow_convnext import FSQFlowPredictorConvNeXt
from Modules.predictor_duration_speaker_context import PredictorDurationSpeakerContext
# Import codec for validation
from Modules.codec_decoder_mel_speaker import MelCodecVocoderSpeaker
warnings.simplefilter("ignore")
logger = get_logger(__name__, log_level="DEBUG")
class Optimizer:
"""Simple optimizer wrapper with linear decay from 1e-5 to 2e-5 over 800k steps."""
def __init__(self, parameters, lr=1e-5, betas=(0.9, 0.999), weight_decay=0.01):
self.optimizer = torch.optim.AdamW(
parameters, lr=lr, betas=betas, weight_decay=weight_decay
)
self.scheduler = None
def set_scheduler(self, total_steps=800000, start_lr=1e-5, final_lr=2e-5):
from torch.optim.lr_scheduler import LambdaLR
def lr_lambda(step):
if step >= total_steps:
return final_lr / start_lr
# Linear decay from start_lr to final_lr
return 1.0 + (final_lr / start_lr - 1.0) * (step / total_steps)
self.scheduler = LambdaLR(self.optimizer, lr_lambda)
def step(self):
self.optimizer.step()
if self.scheduler is not None:
self.scheduler.step()
def zero_grad(self):
self.optimizer.zero_grad()
def get_lr(self):
return self.optimizer.param_groups[0]['lr']
def build_codec_model(config, device):
"""Build mel codec model for validation (decoding predicted tokens)."""
model_params = config.get('model_params', {})
decoder_params = model_params.get('decoder', {})
codec = MelCodecVocoderSpeaker(
num_speakers=model_params.get('num_speakers', model_params.get('n_speakers', 11)),
speaker_dim=model_params.get('speaker_dim', 128),
n_mels=model_params.get('n_mels', 80),
latent_dim=model_params.get('latent_dim', 512),
hidden_dim=model_params.get('hidden_dim', 512),
codec_strides=model_params.get('codec_strides', [2, 2]),
fsq_levels=model_params.get('fsq_levels', [4] * 6),
upsample_rates=decoder_params.get('upsample_rates', model_params.get('upsample_rates', [7, 9])),
gen_istft_n_fft=decoder_params.get('gen_istft_n_fft', model_params.get('gen_istft_n_fft', 28)),
gen_istft_hop_size=decoder_params.get('gen_istft_hop_size', model_params.get('gen_istft_hop_size', 7)),
sample_rate=44100,
n_phonemes=model_params.get('n_phonemes', 178),
num_languages=model_params.get('num_languages', 10),
language_dim=model_params.get('language_dim', 64),
)
return codec.to(device)
@click.command()
@click.option('-p', '--config_path', default='Configs/config_fsq_flow_convnext.yml', type=str)
@click.option('--codec_config_path', default='Configs/config_codec_mel_speaker.yml', type=str)
@click.option('--pretrained_flow_path', default='', type=str, help="Path to pretrained flow predictor checkpoint")
def main(config_path: str, codec_config_path: str, pretrained_flow_path: str):
# Load configs
config = yaml.safe_load(open(config_path, "r"))
codec_config = yaml.safe_load(open(codec_config_path, "r")) if osp.exists(codec_config_path) else {}
# Override log dir for duration training to avoid overwriting flow training
log_dir = config["log_dir"] + "_duration"
os.makedirs(log_dir, exist_ok=True)
shutil.copy(config_path, osp.join(log_dir, osp.basename(config_path)))
# Accelerator setup
ddp_kwargs = DistributedDataParallelKwargs(find_unused_parameters=True)
accelerator = Accelerator(
project_dir=log_dir,
split_batches=True,
kwargs_handlers=[ddp_kwargs],
mixed_precision="bf16",
)
device = accelerator.device
if accelerator.is_main_process:
writer = SummaryWriter(osp.join(log_dir, "tensorboard"))
# Logging
if accelerator.is_main_process:
py_logger = logger.logger
py_logger.setLevel(logging.DEBUG)
py_logger.propagate = False
formatter = logging.Formatter("%(levelname)s:%(asctime)s: %(message)s")
if not py_logger.handlers:
console_handler = logging.StreamHandler(sys.stdout)
console_handler.setLevel(logging.INFO)
console_handler.setFormatter(formatter)
py_logger.addHandler(console_handler)
log_file_path = osp.abspath(osp.join(log_dir, "train.log"))
file_handler = logging.FileHandler(log_file_path)
file_handler.setLevel(logging.DEBUG)
file_handler.setFormatter(formatter)
py_logger.addHandler(file_handler)
# ==========================================================================
# CONFIG PARAMETERS
# ==========================================================================
batch_size = int(config.get("batch_size", 16))
epochs = int(config.get("epochs", 100))
save_freq = int(config.get("save_freq", 2))
log_interval = int(config.get("log_interval", 20))
val_interval = int(config.get("val_interval", 50000))
# Data params
data_params = config.get("data_params", {})
train_path = data_params.get("train_data", "Data/train_list.txt")
val_path = data_params.get("val_data", "Data/val_list.txt")
min_length = int(data_params.get("min_length", 50))
n_mels = int(data_params.get("n_mels", 80))
sample_rate = int(config.get("sample_rate", 44100))
# Model params
model_params = config.get("model_params", {})
vocab_size = int(model_params.get("vocab_size", 178))
text_dim = int(model_params.get("text_dim", 512))
text_num_layers = int(model_params.get("text_num_layers", 6))
n_speakers = int(model_params.get("n_speakers", 11))
speaker_dim = int(model_params.get("speaker_dim", 128))
num_languages = int(model_params.get("num_languages", 10))
language_dim = int(model_params.get("language_dim", 64))
hidden_dim = int(model_params.get("hidden_dim", 512))
flow_num_layers = int(model_params.get("flow_num_layers", 8))
flow_num_heads = int(model_params.get("flow_num_heads", 8))
duration_n_flows = int(model_params.get("duration_n_flows", 4))
fsq_levels = model_params.get("fsq_levels", [4, 4, 4, 4, 4, 4])
codec_stride = int(model_params.get("codec_stride", 2))
n_phonemes = int(model_params.get("n_phonemes", 178))
# Training params
lr = float(config.get("lr", 1e-5))
lr_decay_steps = int(config.get("lr_decay_steps", 800000))
lr_final = float(config.get("lr_final", 2e-5))
duration_pred_weight = float(config.get("duration_pred_weight", 1.0)) # Focus on this
# Paths
alignment_path_train = config.get("alignment_train", "../data_preparation_v2/alignments_train.safetensors")
alignment_path_val = config.get("alignment_val", "../data_preparation_v2/alignments_val.safetensors")
z_continuous_dir_train = config.get("z_continuous_train", "precomputed_tokens/train/z_continuous")
z_continuous_dir_val = config.get("z_continuous_val", "precomputed_tokens/val/z_continuous")
# ==========================================================================
# DATA LOADING
# ==========================================================================
train_list, val_list = get_data_path_list(train_path, val_path)
train_loader = build_fsq_flow_dataloader(
train_list,
z_continuous_dir=z_continuous_dir_train,
alignment_path=alignment_path_train,
validation=False,
min_length=min_length,
batch_size=batch_size,
num_workers=4,
device=device,
balance_speakers=True,
)
val_loader = build_fsq_flow_dataloader(
val_list,
z_continuous_dir=z_continuous_dir_val,
alignment_path=alignment_path_val,
validation=True,
min_length=min_length,
batch_size=1,
num_workers=0,
device=device,
balance_speakers=False,
)
# ==========================================================================
# BUILD MODEL
# ==========================================================================
logger.info("Building models...")
# 1. Flow Predictor (built first so we can share its ConvNeXt text encoder)
flow_predictor = FSQFlowPredictorConvNeXt(
vocab_size=vocab_size,
text_dim=text_dim,
text_num_layers=text_num_layers,
text_kernel_size=7,
num_languages=num_languages,
language_dim=language_dim,
n_speakers=n_speakers,
speaker_dim=speaker_dim,
fsq_levels=fsq_levels,
hidden_dim=hidden_dim,
cond_layers=flow_num_layers,
flow_num_heads=flow_num_heads,
n_phonemes=n_phonemes,
dropout=0.1,
codec_stride=codec_stride,
pad_token_id=0,
)
# Load pretrained flow predictor
if pretrained_flow_path and osp.exists(pretrained_flow_path):
logger.info(f"Loading pretrained flow predictor from {pretrained_flow_path}")
state = torch.load(pretrained_flow_path, map_location='cpu')
if "predictor" in state:
flow_predictor.load_state_dict(state["predictor"], strict=False)
else:
flow_predictor.load_state_dict(state, strict=False)
else:
logger.warning(f"No pretrained flow predictor path provided or found at {pretrained_flow_path}. Validation audio will be random noise.")
# Freeze entire flow predictor, then unfreeze only the text encoder so it
# receives gradients from the duration loss (joint training of shared encoder).
flow_predictor.eval()
for p in flow_predictor.parameters():
p.requires_grad = False
for p in flow_predictor.text_encoder.parameters():
p.requires_grad = True
flow_predictor.to(device)
# 2. Duration Predictor (TRAINABLE) — shares ConvNeXt text encoder with flow predictor
duration_predictor = PredictorDurationSpeakerContext(
text_encoder=flow_predictor.text_encoder,
vocab_size=vocab_size,
n_speakers=n_speakers,
speaker_emb_dim=speaker_dim,
filter_channels=hidden_dim,
p_dropout=0.1,
n_flows=duration_n_flows,
pad_token_id=0,
)
# 3. Codec (FROZEN - for validation)
codec = None
codec_ckpt = config.get("pretrained_codec", "")
if codec_ckpt and osp.exists(codec_ckpt):
print(f"Loading codec from {codec_ckpt}")
codec = build_codec_model(codec_config, device)
state = torch.load(codec_ckpt, map_location='cpu', weights_only=False)
if 'net' in state:
codec.load_state_dict(state['net'].get('codec', state['net']), strict=False)
else:
codec.load_state_dict(state, strict=False)
codec.eval()
for p in codec.parameters():
p.requires_grad = False
# Optimizer — covers duration predictor head + shared text encoder
# (text encoder params are included because it is registered as duration_predictor.text_encoder)
optimizer = Optimizer(
list(duration_predictor.parameters()),
lr=lr,
)
# Prepare with accelerator
duration_predictor = accelerator.prepare(duration_predictor)
optimizer.optimizer = accelerator.prepare(optimizer.optimizer)
# Flow and codec are already on device (if not wrapped) but let's make sure
# (Since they are frozen, we don't need to wrap them with DDP usually unless we need gradients, which we don't)
# However, for inference in validation, being on the right device matters.
duration_predictor = duration_predictor.to(device)
optimizer.set_scheduler(total_steps=lr_decay_steps, start_lr=lr, final_lr=lr_final)
if optimizer.scheduler is not None:
optimizer.scheduler = accelerator.prepare(optimizer.scheduler)
# Load checkpoint if exists
start_epoch = 0
iters = 0
best_val_loss = float('inf')
# Try to resume duration predictor training
potential_ckpt = osp.join(log_dir, "latest_model.pt")
if osp.exists(potential_ckpt):
logger.info(f"Resuming from {potential_ckpt}")
state = torch.load(potential_ckpt, map_location='cpu')
if "duration_predictor" in state:
duration_predictor.load_state_dict(state["duration_predictor"])
if "optimizer" in state:
optimizer.optimizer.load_state_dict(state["optimizer"])
start_epoch = state.get("epoch", 0)
iters = state.get("iters", 0)
best_val_loss = state.get("best_val_loss", float('inf'))
# ==========================================================================
# TRAINING LOOP
# ==========================================================================
logger.info(f"Starting training from epoch {start_epoch}...")
for epoch in range(start_epoch, epochs):
duration_predictor.train()
epoch_loss = 0.0
num_batches = 0
for batch_idx, batch in enumerate(train_loader):
optimizer.zero_grad()
# Unpack batch
speakers = batch[1].to(device)
langs = batch[2].to(device)
# We don't need fsq/codec stuff for training duration
tgt_texts = batch[5].to(device) # [B, T_text]
tgt_lens = batch[6].to(device)
ctx_texts = batch[7].to(device)
ctx_lens = batch[8].to(device)
durations = batch[9].to(device) # [B, T_dur]
token_start_indices = batch[10].to(device)
token_end_indices = batch[11].to(device)
phonemes = tgt_texts
phoneme_lengths = tgt_lens
dur_for_model = durations
dur_output = duration_predictor(
x_tokens=phonemes,
x_lengths=phoneme_lengths,
w_durations=dur_for_model,
speaker_ids=speakers,
language_ids=langs,
ctx_tokens=ctx_texts,
ctx_lens=ctx_lens,
token_start_indices=token_start_indices,
token_end_indices=token_end_indices,
)
loss_dur_pred = dur_output['loss_dur']
loss_total = duration_pred_weight * loss_dur_pred
accelerator.backward(loss_total)
optimizer.step()
epoch_loss += loss_dur_pred.item()
num_batches += 1
iters += 1
if accelerator.is_main_process and batch_idx % log_interval == 0:
writer.add_scalar("train/loss_dur_pred", loss_dur_pred.item(), iters)
writer.add_scalar("train/lr", optimizer.get_lr(), iters)
logger.info(
f"Epoch [{epoch+1}/{epochs}] Batch [{batch_idx}/{len(train_loader)}] "
f"Loss: {loss_dur_pred.item():.4f} LR: {optimizer.get_lr():.6f}"
)
# Validation
if accelerator.is_main_process and iters % val_interval == 0:
val_loss = validate(
duration_predictor, flow_predictor, val_loader,
codec_stride, device, writer, iters, codec,
duration_pred_weight=duration_pred_weight,
sample_rate=sample_rate, num_audio_samples=10
)
if val_loss < best_val_loss:
best_val_loss = val_loss
save_checkpoint(
duration_predictor, optimizer, epoch, iters, best_val_loss,
osp.join(log_dir, "best_model.pt")
)
logger.info(f"New best model saved! Val Loss: {val_loss:.4f}")
duration_predictor.train()
# Save epoch checkpoint
if accelerator.is_main_process and (epoch + 1) % save_freq == 0:
save_checkpoint(
duration_predictor, optimizer, epoch, iters, best_val_loss,
osp.join(log_dir, f"model_epoch_{epoch+1}.pt")
)
save_checkpoint(
duration_predictor, optimizer, epoch, iters, best_val_loss,
osp.join(log_dir, "latest_model.pt")
)
if accelerator.is_main_process:
writer.close()
logger.info("Training complete!")
def validate(
duration_predictor,
flow_predictor,
val_loader,
codec_stride,
device,
writer,
iters,
codec=None,
duration_pred_weight=1.0,
sample_rate=44100,
num_audio_samples=5,
):
"""Run validation and full OOD inference."""
duration_predictor.eval()
total_loss_dur_pred = 0.0
num_samples = 0
audio_samples_generated = 0
with torch.no_grad():
for batch_idx, batch in enumerate(val_loader):
if batch_idx >= 50: break
speakers = batch[1].to(device)
langs = batch[2].to(device)
tgt_texts = batch[5].to(device)
tgt_lens = batch[6].to(device)
ctx_texts = batch[7].to(device)
ctx_lens = batch[8].to(device)
durations = batch[9].to(device)
token_start_indices = batch[10].to(device)
token_end_indices = batch[11].to(device)
phonemes = tgt_texts
phoneme_lengths = tgt_lens
dur_for_model = durations
dur_output = duration_predictor(
x_tokens=phonemes,
x_lengths=phoneme_lengths,
w_durations=dur_for_model,
speaker_ids=speakers,
language_ids=langs,
ctx_tokens=ctx_texts,
ctx_lens=ctx_lens,
token_start_indices=token_start_indices,
token_end_indices=token_end_indices,
)
loss_dur_pred = dur_output['loss_dur']
total_loss_dur_pred += loss_dur_pred.item()
num_samples += 1
# GENERATE AUDIO (Full OOD Pipeline)
if codec is not None and flow_predictor is not None and audio_samples_generated < num_audio_samples:
try:
B = speakers.shape[0]
unwrapped_dur_pred = getattr(duration_predictor, 'module', duration_predictor)
# 1. Predict Duration
pred_dur_ood, _, _ = unwrapped_dur_pred.infer(
x_tokens=phonemes,
x_lengths=phoneme_lengths,
speaker_ids=speakers,
language_ids=langs,
ctx_tokens=ctx_texts,
ctx_lens=ctx_lens,
token_start_indices=token_start_indices,
token_end_indices=token_end_indices,
)
# 2. Predict Flow Embeddings (using predicted duration)
inference_out_ood = flow_predictor.inference(
phonemes=phonemes,
phoneme_lengths=phoneme_lengths,
durations=pred_dur_ood,
speaker_ids=speakers,
language_ids=langs,
n_steps=20,
temperature=0.6,
ctx_tokens=ctx_texts,
ctx_lens=ctx_lens,
token_start_indices=token_start_indices,
token_end_indices=token_end_indices,
)
pred_embeddings_ood = inference_out_ood['pred_embeddings']
for b in range(min(B, num_audio_samples - audio_samples_generated)):
speaker_id = speakers[b:b+1]
lang_id = langs[b:b+1]
# Get lengths
frame_len_ood = pred_dur_ood[b].sum().long().item()
codec_len_ood = max(1, (frame_len_ood + codec_stride - 1) // codec_stride)
z_pred_ood = pred_embeddings_ood[b:b+1, :codec_len_ood]
# 3. Decode Audio
wav_pred_ood, _, _ = codec.decode_from_continuous(
z_pred_ood, speaker_id, lang_id, f0=None
)
writer.add_audio(
f'val/ood_sample_{audio_samples_generated}_spk{speaker_id.item()}',
wav_pred_ood.cpu().numpy().squeeze(),
iters,
sample_rate=sample_rate
)
audio_samples_generated += 1
except Exception as e:
logger.warning(f"Audio gen failed: {e}")
avg_loss_dur = total_loss_dur_pred / max(num_samples, 1)
writer.add_scalar("val/loss_dur_pred", avg_loss_dur, iters)
logger.info(f"Validation Loss (Duration): {avg_loss_dur:.4f}")
return avg_loss_dur
def save_checkpoint(duration_predictor, optimizer, epoch, iters, best_val_loss, path):
state = {
"duration_predictor": duration_predictor.state_dict(),
"optimizer": optimizer.optimizer.state_dict(),
"epoch": epoch + 1,
"iters": iters,
"best_val_loss": best_val_loss,
}
torch.save(state, path)
logger.info(f"Checkpoint saved to {path}")
if __name__ == "__main__":
main()
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