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run_parler_tts_training.py
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run_parler_tts_training.py
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#!/usr/bin/env python
# coding=utf-8
# Copyright 2024 The HuggingFace Inc. team. All rights reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
""" Train Parler-TTS using 🤗 Accelerate"""
import logging
import os
import re
import sys
import time
import math
import contextlib
from multiprocess import set_start_method
from datetime import timedelta
import inspect
from tqdm import tqdm
from pathlib import Path
import torch
from torch.utils.data import DataLoader
import datasets
from datasets import DatasetDict, Dataset, IterableDataset, concatenate_datasets
from huggingface_hub import HfApi
import transformers
from transformers import AutoFeatureExtractor, AutoTokenizer, HfArgumentParser
from transformers.trainer_pt_utils import LengthGroupedSampler
from transformers.optimization import get_scheduler
from transformers.utils import send_example_telemetry
from accelerate import Accelerator, skip_first_batches
from accelerate.utils import set_seed, AutocastKwargs, InitProcessGroupKwargs, TorchDynamoPlugin, DistributedDataParallelKwargs
from accelerate.utils.memory import release_memory
from parler_tts import (
ParlerTTSConfig,
ParlerTTSForConditionalGeneration,
build_delay_pattern_mask,
)
from training.utils import (
get_last_checkpoint,
rotate_checkpoints,
log_pred,
log_metric,
load_all_codec_checkpoints,
save_codec_checkpoint,
get_last_codec_checkpoint_step,
)
from training.arguments import ModelArguments, DataTrainingArguments, ParlerTTSTrainingArguments
from training.data import load_multiple_datasets, DataCollatorParlerTTSWithPadding, DataCollatorEncodecWithPadding
from training.eval import clap_similarity, wer, si_sdr
logger = logging.getLogger(__name__)
def main():
# See all possible arguments in src/transformers/training_args.py
# or by passing the --help flag to this script.
# We now keep distinct sets of args, for a cleaner separation of concerns.
parser = HfArgumentParser((ModelArguments, DataTrainingArguments, ParlerTTSTrainingArguments))
if len(sys.argv) == 2 and sys.argv[1].endswith(".json"):
# If we pass only one argument to the script and it's the path to a json file,
# let's parse it to get our arguments.
model_args, data_args, training_args = parser.parse_json_file(json_file=os.path.abspath(sys.argv[1]))
else:
model_args, data_args, training_args = parser.parse_args_into_dataclasses()
# Sending telemetry. Tracking the example usage helps us better allocate resources to maintain them. The
# information sent is the one passed as arguments along with your Python/PyTorch versions.
send_example_telemetry("run_parler_tts", model_args, data_args)
if training_args.dtype == "float16":
mixed_precision = "fp16"
torch_dtype = torch.float16
elif training_args.dtype == "bfloat16":
mixed_precision = "bf16"
torch_dtype = torch.bfloat16
else:
mixed_precision = "no"
torch_dtype = torch.float32
if data_args.pad_to_max_length and (
data_args.max_duration_in_seconds is None
or data_args.max_prompt_token_length is None
or data_args.max_description_token_length is None
):
raise ValueError(
"`pad_to_max_length` is `True` but one of the following parameters has not been set: `max_duration_in_seconds`, `max_prompt_token_length`, `max_description_token_length`"
)
padding = "max_length" if data_args.pad_to_max_length else "longest"
####### A. Preparation
kwargs_handlers = [InitProcessGroupKwargs(timeout=timedelta(minutes=120)), DistributedDataParallelKwargs(find_unused_parameters=False)]
accelerator = Accelerator(
gradient_accumulation_steps=training_args.gradient_accumulation_steps,
mixed_precision=mixed_precision,
log_with=training_args.report_to,
project_dir=training_args.output_dir,
kwargs_handlers=kwargs_handlers,
)
accelerator.init_trackers(
project_name=data_args.wandb_project,
config={
"learning_rate": training_args.learning_rate,
"model_name_or_path": model_args.model_name_or_path,
"num_train_epochs": training_args.num_train_epochs,
"gradient_accumulation_steps": training_args.gradient_accumulation_steps,
"per_device_train_batch_size": training_args.per_device_train_batch_size,
"global_batch_size": training_args.per_device_train_batch_size * accelerator.num_processes,
"mixed_precision": mixed_precision,
"lr_scheduler_type": training_args.lr_scheduler_type,
"warmup_steps": training_args.warmup_steps,
"freeze_text_encoder": model_args.freeze_text_encoder,
"max_duration_in_seconds": data_args.max_duration_in_seconds,
"weight_decay": training_args.weight_decay,
"adam_beta1": training_args.adam_beta1,
"adam_beta2": training_args.adam_beta2,
"temperature": model_args.temperature,
},
init_kwargs={"wandb": {"name": data_args.wandb_run_name}} if data_args.wandb_run_name else {},
)
# Detecting last checkpoint and eventually continue from last checkpoint
last_checkpoint = None
if os.path.isdir(training_args.output_dir) and training_args.do_train and not training_args.overwrite_output_dir:
last_checkpoint = get_last_checkpoint(training_args.output_dir)
if last_checkpoint is None and len(os.listdir(training_args.output_dir)) > 0:
raise ValueError(
f"Output directory ({training_args.output_dir}) already exists and is not empty. "
"Use --overwrite_output_dir to overcome."
)
elif last_checkpoint is not None and training_args.resume_from_checkpoint is None:
logger.info(
f"Checkpoint detected, resuming training at {last_checkpoint}. To avoid this behavior, change "
"the `--output_dir` or add `--overwrite_output_dir` to train from scratch."
)
# Setup logging
logging.basicConfig(
format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
datefmt="%m/%d/%Y %H:%M:%S",
handlers=[logging.StreamHandler(sys.stdout)],
)
logger.setLevel(logging.INFO if accelerator.is_main_process else logging.WARN)
# Log a small summary on each proces
logger.warning(
f"Process rank: {training_args.local_rank}, device: {training_args.device}, n_gpu: {training_args.n_gpu}, "
f"distributed training: {training_args.parallel_mode.value == 'distributed'}, 16-bits training: {training_args.fp16}"
)
# Set the verbosity to info of the Transformers logger (on main process only)
if accelerator.is_local_main_process:
datasets.utils.logging.set_verbosity_warning()
transformers.utils.logging.set_verbosity_info()
else:
datasets.utils.logging.set_verbosity_error()
transformers.utils.logging.set_verbosity_error()
logger.info("Training/evaluation parameters %s", training_args)
# Set seed before initializing model.
set_seed(training_args.seed)
num_workers = data_args.preprocessing_num_workers
# 1. First, lett's instantiate the feature extractor, tokenizers and model
# Note for distributed training, the .from_pretrained methods guarantee that only
# one local process can concurrently download model & vocab.
# load feature extractor
feature_extractor = AutoFeatureExtractor.from_pretrained(
model_args.feature_extractor_name or model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
token=data_args.token,
trust_remote_code=data_args.trust_remote_code,
)
sampling_rate = feature_extractor.sampling_rate
# load prompt tokenizer
prompt_tokenizer = AutoTokenizer.from_pretrained(
model_args.prompt_tokenizer_name or model_args.description_tokenizer_name or model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
token=data_args.token,
trust_remote_code=data_args.trust_remote_code,
use_fast=model_args.use_fast_tokenizer,
padding_side=model_args.prompt_padding_side,
)
# load description tokenizer
description_tokenizer = AutoTokenizer.from_pretrained(
model_args.description_tokenizer_name or model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
token=data_args.token,
trust_remote_code=data_args.trust_remote_code,
use_fast=model_args.use_fast_tokenizer,
)
if model_args.use_fast_tokenizer:
logger.warning(
"Disabling fast tokenizer warning: https://github.com/huggingface/transformers/blob/main/src/transformers/tokenization_utils_base.py#L3231-L3235"
)
prompt_tokenizer.deprecation_warnings["Asking-to-pad-a-fast-tokenizer"] = True
description_tokenizer.deprecation_warnings["Asking-to-pad-a-fast-tokenizer"] = True
# 2. Now, let's load the dataset
if data_args.save_to_disk is not None:
os.makedirs(data_args.save_to_disk, exist_ok=True)
# assume that the dataset has been saved to `save_to_disk` if the latter is not empty
dataset_was_precomputed = len(os.listdir(data_args.save_to_disk)) > 0
if dataset_was_precomputed:
with accelerator.local_main_process_first():
vectorized_datasets = datasets.load_from_disk(data_args.save_to_disk)
else:
raw_datasets = DatasetDict()
columns_to_keep = {
"target_audio_column_name": data_args.target_audio_column_name,
"prompt_column_name": data_args.prompt_column_name,
}
if data_args.description_column_name is not None:
columns_to_keep["description_column_name"] = data_args.description_column_name
if training_args.do_train:
raw_datasets["train"] = load_multiple_datasets(
accelerator,
data_args.train_dataset_name,
data_args.train_dataset_config_name,
metadata_dataset_names=data_args.train_metadata_dataset_name,
splits=data_args.train_split_name,
dataset_samples=data_args.train_dataset_samples,
seed=training_args.seed,
cache_dir=model_args.cache_dir,
num_proc=data_args.preprocessing_num_workers,
id_column_name=data_args.id_column_name,
columns_to_keep=columns_to_keep.values(),
prompt_column_name=data_args.prompt_column_name,
audio_column_name=data_args.target_audio_column_name,
sampling_rate=sampling_rate,
logger=logger,
# streaming=data_args.streaming, TODO(SG): optionally enable streaming mode
)
for key in columns_to_keep:
if columns_to_keep[key] not in raw_datasets["train"].column_names:
raise ValueError(
f"--{key} '{columns_to_keep[key]}' not found in dataset '{data_args.train_dataset_name}'."
f" Make sure to set `--{key}` to the correct audio column - one of"
f" {', '.join(raw_datasets['train'].column_names)}."
)
if data_args.max_train_samples is not None:
raw_datasets["train"] = raw_datasets["train"].select(range(data_args.max_train_samples))
if training_args.do_eval:
raw_datasets["eval"] = load_multiple_datasets(
accelerator,
data_args.eval_dataset_name if data_args.eval_dataset_name else data_args.train_dataset_name,
data_args.eval_dataset_config_name
if data_args.eval_dataset_config_name
else data_args.train_dataset_config_name,
metadata_dataset_names=data_args.eval_metadata_dataset_name,
splits=data_args.eval_split_name,
cache_dir=model_args.cache_dir,
num_proc=data_args.preprocessing_num_workers,
id_column_name=data_args.id_column_name,
columns_to_keep=columns_to_keep.values(),
prompt_column_name=data_args.prompt_column_name,
audio_column_name=data_args.target_audio_column_name,
sampling_rate=sampling_rate,
logger=logger,
# streaming=data_args.streaming, TODO(SG): optionally enable streaming mode
)
if data_args.max_eval_samples is not None:
with accelerator.local_main_process_first():
raw_datasets["eval"] = (
raw_datasets["eval"].shuffle(seed=training_args.seed).select(range(data_args.max_eval_samples))
)
# 3. Next, let's load the config.
config = ParlerTTSConfig.from_pretrained(
model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
token=data_args.token,
trust_remote_code=data_args.trust_remote_code,
)
if training_args.codebook_weights is not None and len(training_args.codebook_weights) != config.decoder.num_codebooks:
raise ValueError(f"`codebook_weights` has length {len(training_args.codebook_weights)} when it should be of length {config.decoder.num_codebooks}.")
# update pad token id and decoder_start_token_id
config.decoder.update(
{
"cross_attention_implementation_strategy": model_args.cross_attention_implementation_strategy
if model_args.cross_attention_implementation_strategy is not None
else None,
"codebook_weights": training_args.codebook_weights if training_args.codebook_weights is not None else config.decoder.codebook_weights
}
)
config.update(
{
"pad_token_id": model_args.pad_token_id if model_args.pad_token_id is not None else config.pad_token_id,
"decoder_start_token_id": model_args.decoder_start_token_id
if model_args.decoder_start_token_id is not None
else config.decoder_start_token_id,
}
)
# create model
model = ParlerTTSForConditionalGeneration.from_pretrained(
model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
config=config,
token=data_args.token,
trust_remote_code=data_args.trust_remote_code,
attn_implementation={"decoder": model_args.attn_implementation, "text_encoder": "eager"},
)
# enable gradient checkpointing if necessary
if training_args.gradient_checkpointing:
model.gradient_checkpointing_enable()
# 4. Now we preprocess the datasets including loading the audio, resampling and normalization
# Thankfully, `datasets` takes care of automatically loading and resampling the audio,
# so that we just need to set the correct target sampling rate and normalize the input
# via the `feature_extractor`
# derive max & min input length for sample rate & max duration
sampling_rate = feature_extractor.sampling_rate
max_target_length = int(data_args.max_duration_in_seconds * sampling_rate)
min_target_length = int(data_args.min_duration_in_seconds * sampling_rate)
target_audio_column_name = data_args.target_audio_column_name
description_column_name = data_args.description_column_name
prompt_column_name = data_args.prompt_column_name
feature_extractor_input_name = feature_extractor.model_input_names[0]
audio_encoder_pad_token_id = config.decoder.pad_token_id
audio_encoder_eos_token_id = config.decoder.eos_token_id
audio_encoder_bos_token_id = model.generation_config.decoder_start_token_id
max_length = model.generation_config.max_length
num_codebooks = model.decoder.config.num_codebooks
bandwidth = model_args.bandwidth
attn_implementation = model_args.attn_implementation
# Freeze Encoders
model.freeze_encoders(model_args.freeze_text_encoder)
# Test all gather - used for warmout and avoiding timeout
logger.debug(str(accelerator.process_index), main_process_only=False, in_order=True)
test_tensor = torch.tensor([accelerator.process_index], device=accelerator.device)
gathered_tensor = accelerator.gather(test_tensor)
print("gathered_tensor", gathered_tensor)
accelerator.wait_for_everyone()
if not dataset_was_precomputed:
# Filter on text length
if description_column_name is not None and data_args.max_text_length is not None:
with accelerator.local_main_process_first():
# filter description that is shorter than max_text_length
raw_datasets = raw_datasets.filter(
lambda x: len(x) < data_args.max_text_length,
num_proc=num_workers,
input_columns=[description_column_name],
)
# Preprocessing the dataset.
# We need to tokenize the texts.
def pass_through_processors(description, prompt):
batch = {}
batch["input_ids"] = description_tokenizer(description.strip())["input_ids"]
batch["prompt_input_ids"] = prompt_tokenizer(prompt.strip())["input_ids"]
return batch
with accelerator.local_main_process_first():
# this is a trick to avoid to rewrite the entire audio column which takes ages
vectorized_datasets = raw_datasets.map(
pass_through_processors,
remove_columns=next(iter(raw_datasets.values())).column_names,
input_columns=[description_column_name, prompt_column_name],
num_proc=num_workers,
desc="preprocess datasets",
)
# We use Accelerate to perform distributed inference
# T5 doesn't support fp16
autocast_kwargs = AutocastKwargs(enabled=(mixed_precision != "fp16"))
# Now we encode the audio labels with encodec.
####### B. Encode audio
logger.info("*** Encode target audio with encodec ***")
# no need to prepare audio_decoder because used for inference without mixed precision
# see: https://huggingface.co/docs/accelerate/main/en/package_reference/accelerator#accelerate.Accelerator.prepare
if training_args.torch_compile:
audio_decoder = accelerator.prepare_model(model.audio_encoder, evaluation_mode=True)
else:
audio_decoder = model.audio_encoder
encoder_data_collator = DataCollatorEncodecWithPadding(
feature_extractor,
audio_column_name=target_audio_column_name,
feature_extractor_input_name=feature_extractor_input_name,
max_length=max_target_length,
padding=padding,
)
encoder_signature = set(inspect.signature(audio_decoder.forward).parameters)
def apply_audio_decoder(batch):
len_audio = batch.pop("len_audio")
audio_decoder.to(batch["input_values"].device).eval()
if bandwidth is not None:
batch["bandwidth"] = bandwidth
elif "num_quantizers" in encoder_signature:
batch["num_quantizers"] = num_codebooks
elif "num_codebooks" in encoder_signature:
batch["num_codebooks"] = num_codebooks
elif "n_quantizers" in encoder_signature:
batch["n_quantizers"] = num_codebooks
with torch.no_grad():
labels = audio_decoder.encode(**batch)["audio_codes"]
output = {}
output["len_audio"] = len_audio
# (1, bsz, codebooks, seq_len) -> (bsz, seq_len, codebooks)
output["labels"] = labels.squeeze(0).transpose(1, 2)
# if `pad_to_max_length`, the maximum corresponding audio length of the current batch is max_duration*sampling_rate
max_length = len_audio.max() if padding != "max_length" else max_target_length
output["ratio"] = torch.ones_like(len_audio) * labels.shape[-1] / max_length
return output
# (1, codebooks, seq_len) where seq_len=1
bos_labels = torch.ones((1, num_codebooks, 1)) * audio_encoder_bos_token_id
def postprocess_dataset(labels):
# (1, codebooks, seq_len)
labels = torch.tensor(labels).unsqueeze(0)
# add bos
labels = torch.cat([bos_labels, labels], dim=-1)
labels, delay_pattern_mask = build_delay_pattern_mask(
labels,
bos_token_id=audio_encoder_bos_token_id,
pad_token_id=audio_encoder_eos_token_id,
max_length=labels.shape[-1] + num_codebooks,
num_codebooks=num_codebooks,
)
# the first ids of the delay pattern mask are precisely labels, we use the rest of the labels mask
# to take care of EOS
# we want labels to look like this:
# - [B, a, b, E, E, E, E]
# - [B, B, c, d, E, E, E]
# - [B, B, B, e, f, E, E]
# - [B, B, B, B, g, h, E]
labels = torch.where(delay_pattern_mask == -1, audio_encoder_eos_token_id, delay_pattern_mask)
# the first timestamp is associated to a row full of BOS, let's get rid of it
# we also remove the last timestampts (full of PAD)
output = {"labels": labels[:, 1:]}
return output
for split in vectorized_datasets:
data_loader = DataLoader(
raw_datasets[split],
batch_size=training_args.audio_encoder_per_device_batch_size,
collate_fn=encoder_data_collator,
num_workers=training_args.dataloader_num_workers,
pin_memory=True,
)
data_loader = accelerator.prepare(data_loader)
total_inference_steps = len(data_loader)
start_step = get_last_codec_checkpoint_step(os.path.join(data_args.temporary_save_to_disk, split))
accelerator.wait_for_everyone()
if start_step > 0:
logger.info(f"Resuming {split} from step {start_step}")
# efficiently skip the first n batches
start_step += 1
data_loader = skip_first_batches(data_loader, start_step)
all_generated_labels = []
all_lens = []
if start_step < total_inference_steps:
for i, batch in enumerate(tqdm(data_loader, disable=not accelerator.is_local_main_process)):
cur_step = start_step + i
generate_labels = apply_audio_decoder(batch)
generate_labels = accelerator.pad_across_processes(generate_labels, dim=1, pad_index=0)
generate_labels = accelerator.gather_for_metrics(generate_labels)
if accelerator.is_main_process:
lab = generate_labels["labels"].cpu().transpose(1, 2).to(torch.int16)
rat = generate_labels["ratio"].cpu().squeeze(1)
lens = generate_labels["len_audio"].cpu().squeeze(1)
lab = [l[:, : int(ratio * length)] for (l, ratio, length) in zip(lab, rat, lens)]
all_generated_labels.extend(lab)
all_lens.extend(lens)
if ((cur_step + 1) % data_args.save_codec_steps == 0) or (
cur_step == total_inference_steps - 1
):
tmp_labels = Dataset.from_dict({"labels": all_generated_labels, "target_length": all_lens})
tmp_labels = tmp_labels.map(
postprocess_dataset,
num_proc=data_args.preprocessing_num_workers, # this one is resource consuming if many processor.
input_columns=["labels"],
desc="Postprocessing labeling",
)
save_codec_checkpoint(
os.path.join(data_args.temporary_save_to_disk, split), tmp_labels, cur_step
)
all_generated_labels = []
all_lens = []
accelerator.wait_for_everyone()
if accelerator.is_main_process and len(all_generated_labels) > 0:
tmp_labels = Dataset.from_dict({"labels": all_generated_labels, "target_length": all_lens})
tmp_labels = tmp_labels.map(
postprocess_dataset,
num_proc=data_args.preprocessing_num_workers, # this one is resource consuming if many processor.
input_columns=["labels"],
desc="Postprocessing labeling",
)
save_codec_checkpoint(os.path.join(data_args.temporary_save_to_disk, split), tmp_labels, cur_step)
all_generated_labels = []
all_lens = []
accelerator.wait_for_everyone()
del all_generated_labels
accelerator.wait_for_everyone()
with accelerator.local_main_process_first():
tmp_labels = load_all_codec_checkpoints(os.path.join(data_args.temporary_save_to_disk, split)).select(
range(len(vectorized_datasets[split]))
)
logger.info(f"Concatenating {split}: {tmp_labels} with {vectorized_datasets[split]}")
vectorized_datasets[split] = concatenate_datasets([vectorized_datasets[split], tmp_labels], axis=1)
accelerator.free_memory()
del generate_labels, all_lens
with accelerator.local_main_process_first():
# NOTE: filtering is done at the end because in the `datasets` library, caching audio files is done after most operations
# caching audio files is time and disk-space consuming, so we want to avoid it at all costs, especially for large (>1Kh) audio datasets.
# That's also why we avoid to concat the processed datasets (vectorized_datasets) with the audio column present in raw_datasets.
def is_audio_in_length_range(length):
return length > min_target_length and length < max_target_length
# filter data that is shorter than min_target_length
vectorized_datasets = vectorized_datasets.filter(
is_audio_in_length_range,
num_proc=num_workers,
input_columns=["target_length"],
)
if description_column_name is not None and data_args.max_description_token_length is not None:
with accelerator.local_main_process_first():
# filter description that is shorter than max_text_length
vectorized_datasets = vectorized_datasets.filter(
lambda x: len(x) < data_args.max_description_token_length,
num_proc=num_workers,
input_columns=["input_ids"],
)
if data_args.max_prompt_token_length is not None:
with accelerator.local_main_process_first():
# filter description that is shorter than max_text_length
vectorized_datasets = vectorized_datasets.filter(
lambda x: len(x) < data_args.max_prompt_token_length,
num_proc=num_workers,
input_columns=["prompt_input_ids"],
)
if data_args.save_to_disk is not None and not dataset_was_precomputed:
if accelerator.is_main_process:
vectorized_datasets.save_to_disk(
data_args.save_to_disk,
num_proc=min(data_args.preprocessing_num_workers, len(vectorized_datasets["eval"]) - 1),
)
accelerator.wait_for_everyone()
logger.info(f"Dataset saved at {data_args.save_to_disk}")
audio_max_length = None
if padding == "max_length":
audio_max_length = max(vectorized_datasets["train"]["target_length"])
with accelerator.local_main_process_first():
max_sample = vectorized_datasets["train"].filter(
lambda x: x == audio_max_length,
num_proc=num_workers,
input_columns=["target_length"],
)
audio_max_length = max([len(l[0]) for l in max_sample["labels"]])
if description_column_name is not None and data_args.max_description_token_length is not None:
with accelerator.local_main_process_first():
# filter description that is shorter than max_text_length
vectorized_datasets = vectorized_datasets.filter(
lambda x: len(x) < data_args.max_description_token_length,
num_proc=num_workers,
input_columns=["input_ids"],
)
if data_args.max_prompt_token_length is not None:
with accelerator.local_main_process_first():
# filter description that is shorter than max_text_length
vectorized_datasets = vectorized_datasets.filter(
lambda x: len(x) < data_args.max_prompt_token_length,
num_proc=num_workers,
input_columns=["prompt_input_ids"],
)
if training_args.group_by_length:
# apply a simple heuristic to take into account audio and text lengths
def add_target_lengths(target_length, prompt, description):
return {"target_length": target_length + len(prompt) + len(description)}
with accelerator.local_main_process_first():
vectorized_datasets = vectorized_datasets.map(
add_target_lengths,
num_proc=num_workers,
input_columns=["target_length", "prompt_input_ids", "input_ids"],
)
# for large datasets it is advised to run the preprocessing on a
# single machine first with ``args.preprocessing_only`` since there will mostly likely
# be a timeout when running the script in distributed mode.
# In a second step ``args.preprocessing_only`` can then be set to `False` to load the
# cached dataset
if data_args.preprocessing_only and data_args.save_to_disk is None:
raise ValueError(
"`preprocessing_only=True` but `save_to_disk` is not set. The latter should indicates where to save the dataset locally."
)
elif data_args.preprocessing_only:
logger.info(f"Data preprocessing finished. Files save at {data_args.save_to_disk}")
return
# 6. Next, we can prepare the training.
# Let's use word CLAP similary and WER metrics as our evaluation metrics,
def compute_metrics(
audios,
descriptions,
prompts,
device="cpu",
compute_clap_similarity_metric=False,
compute_noise_level_metric=False,
noise_level_to_compute_clean_wer=None,
):
results = {}
input_ids = descriptions
texts = description_tokenizer.batch_decode(input_ids, skip_special_tokens=True)
prompts = prompt_tokenizer.batch_decode(prompts, skip_special_tokens=True)
audios = [a.float().cpu().numpy() for a in audios]
if compute_clap_similarity_metric:
clap_score = clap_similarity(
model_args.clap_model_name_or_path, texts, audios, device, input_sampling_rate=sampling_rate
)
results["clap"] = clap_score
si_sdr_measures = None
if compute_noise_level_metric:
si_sdr_measures = si_sdr(audios, device, input_sampling_rate=sampling_rate)
word_error, transcriptions, clean_word_error, noisy_word_error, percent_clean_samples = wer(
model_args.asr_model_name_or_path,
prompts,
audios,
device,
training_args.per_device_eval_batch_size,
sampling_rate,
noise_level_to_compute_clean_wer,
si_sdr_measures,
)
results["wer"] = word_error
if clean_word_error is not None:
results["clean_wer"] = clean_word_error
results["noisy_word_error"] = noisy_word_error
results["percent_clean_samples"] = percent_clean_samples
return results, texts, prompts, audios, transcriptions, si_sdr_measures
# Define Training Schedule
# Store some constants
per_device_train_batch_size = int(training_args.per_device_train_batch_size)
train_batch_size = per_device_train_batch_size * accelerator.num_processes
gradient_accumulation_steps = int(training_args.gradient_accumulation_steps)
per_device_eval_batch_size = int(training_args.per_device_eval_batch_size)
if training_args.max_steps < 0:
num_epochs = int(training_args.num_train_epochs)
steps_per_epoch = len(vectorized_datasets["train"]) // (train_batch_size * gradient_accumulation_steps)
total_train_steps = steps_per_epoch * num_epochs
elif training_args.max_steps > 0:
logger.info("max_steps is given, it will override any value given in num_train_epochs")
total_train_steps = int(training_args.max_steps)
# Setting a very large number of epochs so we go as many times as necessary over the iterator.
num_epochs = sys.maxsize
steps_per_epoch = total_train_steps
if training_args.eval_steps is None:
logger.info(f"eval_steps is not set, evaluating at the end of each epoch")
eval_steps = steps_per_epoch
else:
eval_steps = training_args.eval_steps
if training_args.eval_generation_steps is None:
eval_generation_steps = eval_steps
else:
eval_generation_steps = training_args.eval_generation_steps
# T5 doesn't support fp16
autocast_kwargs = AutocastKwargs(enabled=(mixed_precision != "fp16"))
# Define optimizer, LR scheduler, collator
optimizer = torch.optim.AdamW(
params=model.parameters(),
lr=training_args.learning_rate,
betas=(training_args.adam_beta1, training_args.adam_beta2),
eps=training_args.adam_epsilon,
weight_decay=training_args.weight_decay,
)
# LR scheduler gets stepped by `num_processes` each time -> account for this in warmup / total steps
lr_scheduler = get_scheduler(
name=training_args.lr_scheduler_type,
optimizer=optimizer,
num_warmup_steps=training_args.get_warmup_steps(total_train_steps) * accelerator.num_processes,
num_training_steps=total_train_steps * accelerator.num_processes,
)
# Instantiate custom data collator
data_collator = DataCollatorParlerTTSWithPadding(
prompt_tokenizer=prompt_tokenizer,
description_tokenizer=description_tokenizer,
pad_to_multiple_of=data_args.pad_to_multiple_of,
padding=padding,
prompt_max_length=data_args.max_prompt_token_length,
description_max_length=data_args.max_description_token_length,
audio_max_length=audio_max_length,
)
# Prepare everything with accelerate
model, optimizer, lr_scheduler = accelerator.prepare(model, optimizer, lr_scheduler)
num_examples = total_train_steps * train_batch_size * gradient_accumulation_steps
logger.info("***** Running training *****")
logger.info(f" Num examples = {num_examples}")
logger.info(" Instantaneous batch size per device =" f" {per_device_train_batch_size}")
logger.info(" Gradient accumulation steps =" f" {gradient_accumulation_steps}")
logger.info(
f" Total train batch size (w. parallel & distributed) = {train_batch_size * gradient_accumulation_steps}"
)
logger.info(f" Total optimization steps = {total_train_steps}")
# ======================== Training ================================
train_time = 0
train_start = time.time()
steps_trained_progress_bar = tqdm(
range(total_train_steps), desc="Train steps ... ", position=0, disable=not accelerator.is_local_main_process
)
continue_training = True
epochs_trained = 0
cur_step = 0
checkpoint = None
if training_args.resume_from_checkpoint is not None:
checkpoint = training_args.resume_from_checkpoint
elif last_checkpoint is not None:
checkpoint = last_checkpoint
if accelerator.is_main_process:
if training_args.push_to_hub:
api = HfApi(token=training_args.hub_token)
# Create repo (repo_name from args or inferred)
repo_name = training_args.hub_model_id
if repo_name is None:
repo_name = Path(training_args.output_dir).absolute().name
repo_id = api.create_repo(repo_name, exist_ok=True).repo_id
with open(os.path.join(training_args.output_dir, ".gitignore"), "w+") as gitignore:
if "wandb" not in gitignore:
gitignore.write("wandb\n")
elif training_args.output_dir is not None:
os.makedirs(training_args.output_dir, exist_ok=True)
accelerator.wait_for_everyone()
# Now save everything to be able to create a single processor later
# make sure all processes wait until data is saved
# only the main process saves them
if accelerator.is_main_process:
# save feature extractor, tokenizer and config
if (
model_args.prompt_tokenizer_name is None
and model_args.description_tokenizer_name
or (model_args.prompt_tokenizer_name == model_args.description_tokenizer_name)
):
prompt_tokenizer.save_pretrained(training_args.output_dir)
else:
logger.warning(
f"Prompt tokenizer ('{model_args.prompt_tokenizer_name}') and description tokenizer ('{model_args.description_tokenizer_name}') are not the same. Saving only the prompt tokenizer."
)
prompt_tokenizer.save_pretrained(training_args.output_dir)
feature_extractor.save_pretrained(training_args.output_dir)
config.save_pretrained(training_args.output_dir)
accelerator.wait_for_everyone()
if checkpoint is not None:
accelerator.load_state(checkpoint)
# Find num steps and epoch from saved state string pattern
pattern = r"checkpoint-(\d+)-epoch-(\d+)"
match = re.search(pattern, checkpoint)
cur_step = int(match.group(1))
epochs_trained = int(match.group(2))
logger.info(" Continuing training from checkpoint, will skip to saved global_step")
logger.info(f" Continuing training from epoch {epochs_trained}")
logger.info(f" Continuing training from global step {cur_step}")
steps_trained_progress_bar.update(cur_step)
for epoch in range(0, epochs_trained):
with accelerator.local_main_process_first():
vectorized_datasets["train"] = vectorized_datasets["train"].shuffle(training_args.seed)
if training_args.max_steps < 0:
# we know exactly the number of steps per epoch, so can skip through the required number of batches
resume_step = (cur_step - epochs_trained * steps_per_epoch) * gradient_accumulation_steps
else:
# Currently we don't know how many steps we've taken in the current epoch
# So we just shuffle the dataset one extra time and start from a fresh epoch
# This is "good enough" for our purposes but not fully correct
resume_step = None
with accelerator.local_main_process_first():
vectorized_datasets["train"] = vectorized_datasets["train"].shuffle(training_args.seed)
else:
resume_step = None
gen_kwargs = {
"do_sample": model_args.do_sample,
"temperature": model_args.temperature,
"max_length": model_args.max_length,
# Because of the delayed pattern mask, generation might stop earlier because of unexpected behaviour
# on the first tokens of the codebooks that are delayed.
# This fix the issue.
"min_new_tokens": num_codebooks + 1,
}
# Define gradient update step fn
def train_step(
batch,
accelerator,
autocast_kwargs,
num_items_in_batch,
gradient_accumulation_steps,
):
if mixed_precision == "fp16":
# fp16 doesn't work with T5-like models
with accelerator.autocast(autocast_handler=autocast_kwargs):
if training_args.parallel_mode.value != "distributed":
encoder_outputs = model.text_encoder(
input_ids=batch.get("input_ids"), attention_mask=batch.get("attention_mask", None)
)
else:
encoder_outputs = model.module.text_encoder(
input_ids=batch.get("input_ids"), attention_mask=batch.get("attention_mask", None)
)
# we optionnally project last_hidden_state to avoid recomputing every time
encoder_hidden_states = encoder_outputs.last_hidden_state
if (
config.text_encoder.hidden_size != config.decoder.hidden_size
and config.decoder.cross_attention_hidden_size is None
):
encoder_hidden_states = (
model.enc_to_dec_proj(encoder_hidden_states)
if training_args.parallel_mode.value != "distributed"
else model.module.enc_to_dec_proj(encoder_hidden_states)
)
if batch.get("attention_mask", None) is not None:
encoder_hidden_states = encoder_hidden_states * batch.get("attention_mask", None)[..., None]
encoder_outputs.last_hidden_state = encoder_hidden_states
batch["encoder_outputs"] = encoder_outputs
outputs = model(**batch, loss_reduction="sum")
# CE (data) loss
ce_loss = (outputs.loss * gradient_accumulation_steps * accelerator.num_processes) / num_items_in_batch
metrics = {"loss": ce_loss}
# per CE loss
per_codebook_losses = outputs.per_codebook_losses
metrics.update({f"codebook_{i}_loss": ((l * gradient_accumulation_steps * accelerator.num_processes) / num_items_in_batch) for (i,l) in enumerate(per_codebook_losses)})
return ce_loss, metrics
# Define eval fn
def eval_step(
batch,
accelerator,
autocast_kwargs,
):
eval_model = model if not training_args.torch_compile else model._orig_mod
if mixed_precision == "fp16":
# fp16 doesn't work with T5-like models
with accelerator.autocast(autocast_handler=autocast_kwargs):
if training_args.parallel_mode.value != "distributed":
encoder_outputs = model.text_encoder(
input_ids=batch.get("input_ids"), attention_mask=batch.get("attention_mask", None)
)
else:
encoder_outputs = model.module.text_encoder(
input_ids=batch.get("input_ids"), attention_mask=batch.get("attention_mask", None)
)
# we optionnally project last_hidden_state to avoid recomputing every time
encoder_hidden_states = encoder_outputs.last_hidden_state
if (
config.text_encoder.hidden_size != config.decoder.hidden_size
and config.decoder.cross_attention_hidden_size is None
):
encoder_hidden_states = (
model.enc_to_dec_proj(encoder_hidden_states)
if training_args.parallel_mode.value != "distributed"
else model.module.enc_to_dec_proj(encoder_hidden_states)
)
if batch.get("attention_mask", None) is not None:
encoder_hidden_states = encoder_hidden_states * batch.get("attention_mask", None)[..., None]
encoder_outputs.last_hidden_state = encoder_hidden_states
batch["encoder_outputs"] = encoder_outputs
with torch.no_grad():
outputs = eval_model(**batch)
# CE (data) loss
ce_loss = outputs.loss
metrics = {"loss": ce_loss}
# per CE loss
per_codebook_losses = outputs.per_codebook_losses
metrics.update({f"codebook_{i}_loss": l for (i,l) in enumerate(per_codebook_losses)})
return metrics
def generate_step(batch, accelerator):
batch.pop("decoder_attention_mask", None)
eval_model = accelerator.unwrap_model(model, keep_fp32_wrapper=True)
if training_args.torch_compile:
# if the model is compiled, we use the original model bc compile is not compatible with .generate
eval_model = model._orig_mod
# since we've might have loaded the weights in fp32, we have to autocast to ensure FA2 weights are in half-precision.
# with accelerator.autocast(autocast_handler=AutocastKwargs(enabled=(attn_implementation=="flash_attention_2"))):
output_audios = eval_model.generate(**batch, **gen_kwargs)
output_audios = accelerator.pad_across_processes(output_audios, dim=1, pad_index=0)
return output_audios
model.train()
total_batched_samples = resume_step if resume_step is not None else 0
for epoch in range(epochs_trained, num_epochs):
with accelerator.local_main_process_first():
vectorized_datasets["train"] = vectorized_datasets["train"].shuffle(training_args.seed)
sampler = None
if training_args.group_by_length:
sampler = LengthGroupedSampler(train_batch_size, lengths=vectorized_datasets["train"]["target_length"])
train_dataloader = DataLoader(
vectorized_datasets["train"],
collate_fn=data_collator,
batch_size=per_device_train_batch_size,