whispertest / run_speech_recognition_seq2seq_streaming.py
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Duplicate from NbAiLab/whisper-large-v2-nob
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#!/usr/bin/env python
# coding=utf-8
# Copyright 2022 The HuggingFace 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.
"""
Fine-tuning the library models for sequence to sequence speech recognition
with 🤗 Datasets' streaming mode.
"""
# You can also adapt this script for your own sequence to sequence speech
# recognition task. Pointers for this are left as comments.
import logging
import os
import sys
from dataclasses import dataclass, field
from typing import Any, Dict, List, Optional, Union
import datasets
import torch
from datasets import DatasetDict, IterableDatasetDict, interleave_datasets, load_dataset
from torch.utils.data import IterableDataset
import evaluate
import transformers
from transformers import (
AutoConfig,
AutoFeatureExtractor,
AutoModelForSpeechSeq2Seq,
AutoProcessor,
AutoTokenizer,
HfArgumentParser,
Seq2SeqTrainer,
Seq2SeqTrainingArguments,
TrainerCallback,
set_seed,
)
from transformers.models.whisper.english_normalizer import BasicTextNormalizer
from transformers.trainer_pt_utils import IterableDatasetShard
from transformers.trainer_utils import get_last_checkpoint, is_main_process
from transformers.utils import check_min_version, send_example_telemetry
from transformers.utils.versions import require_version
# Will error if the minimal version of Transformers is not installed. Remove at your own risks.
check_min_version("4.25.0.dev0")
require_version("datasets>=1.18.2", "To fix: pip install -r examples/pytorch/speech-recognition/requirements.txt")
logger = logging.getLogger(__name__)
@dataclass
class ModelArguments:
"""
Arguments pertaining to which model/config/tokenizer we are going to fine-tune from.
"""
model_name_or_path: str = field(
metadata={"help": "Path to pretrained model or model identifier from huggingface.co/models"}
)
config_name: Optional[str] = field(
default=None, metadata={"help": "Pretrained config name or path if not the same as model_name"}
)
tokenizer_name: Optional[str] = field(
default=None, metadata={"help": "Pretrained tokenizer name or path if not the same as model_name"}
)
feature_extractor_name: Optional[str] = field(
default=None, metadata={"help": "feature extractor name or path if not the same as model_name"}
)
cache_dir: Optional[str] = field(
default=None,
metadata={"help": "Where to store the pretrained models downloaded from huggingface.co"},
)
use_fast_tokenizer: bool = field(
default=True,
metadata={"help": "Whether to use one of the fast tokenizer (backed by the tokenizers library) or not."},
)
model_revision: str = field(
default="main",
metadata={"help": "The specific model version to use (can be a branch name, tag name or commit id)."},
)
use_auth_token: bool = field(
default=False,
metadata={
"help": (
"Will use the token generated when running `huggingface-cli login` (necessary to use this script "
"with private models)."
)
},
)
freeze_feature_encoder: bool = field(
default=True, metadata={"help": "Whether to freeze the feature encoder layers of the model."}
)
freeze_encoder: bool = field(
default=False, metadata={"help": "Whether to freeze the entire encoder of the seq2seq model."}
)
forced_decoder_ids: List[List[int]] = field(
default=None,
metadata={
"help": (
"A list of pairs of integers which indicates a mapping from generation indices to token indices "
"that will be forced before sampling. For example, [[0, 123]] means the first generated token "
"will always be a token of index 123."
)
},
)
suppress_tokens: List[int] = field(
default=None, metadata={"help": "A list of tokens that will be suppressed at generation."}
)
model_index_name: str = field(default=None, metadata={"help": "Pretty name for the model card."})
@dataclass
class DataTrainingArguments:
"""
Arguments pertaining to what data we are going to input our model for training and eval.
"""
dataset_name: str = field(
default=None, metadata={"help": "The name of the dataset to use (via the datasets library)."}
)
dataset_config_name: Optional[str] = field(
default=None, metadata={"help": "The configuration name of the dataset to use (via the datasets library)."}
)
text_column: Optional[str] = field(
default=None,
metadata={"help": "The name of the column in the datasets containing the full texts (for summarization)."},
)
max_train_samples: Optional[int] = field(
default=None,
metadata={
"help": (
"For debugging purposes or quicker training, truncate the number of training examples to this "
"value if set."
)
},
)
max_eval_samples: Optional[int] = field(
default=None,
metadata={
"help": (
"For debugging purposes or quicker training, truncate the number of evaluation examples to this "
"value if set."
)
},
)
audio_column_name: str = field(
default="audio",
metadata={"help": "The name of the dataset column containing the audio data. Defaults to 'audio'"},
)
text_column_name: str = field(
default="text",
metadata={"help": "The name of the dataset column containing the text data. Defaults to 'text'"},
)
max_duration_in_seconds: float = field(
default=20.0,
metadata={
"help": (
"Truncate audio files that are longer than `max_duration_in_seconds` seconds to"
" 'max_duration_in_seconds`"
)
},
)
min_duration_in_seconds: float = field(
default=0.0, metadata={"help": "Filter audio files that are shorter than `min_duration_in_seconds` seconds"}
)
train_split_name: str = field(
default="train",
metadata={
"help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'"
},
)
eval_split_name: str = field(
default="test",
metadata={
"help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'"
},
)
do_lower_case: bool = field(
default=False,
metadata={"help": "Whether the target text should be lower cased."},
)
do_remove_punctuation: bool = field(
default=False,
metadata={"help": "Whether the target text should be striped of punctuation."},
)
do_normalize_eval: bool = field(
default=True,
metadata={"help": "Whether to normalise the references and predictions in the eval WER calculation."},
)
language: str = field(
default=None,
metadata={
"help": (
"Language for multilingual fine-tuning. This argument should be set for multilingual fine-tuning "
"only. For English speech recognition, it should be set to `None`."
)
},
)
task: str = field(
default="transcribe",
metadata={"help": "Task, either `transcribe` for speech recognition or `translate` for speech translation."},
)
shuffle_buffer_size: Optional[int] = field(
default=500,
metadata={
"help": (
"The number of streamed examples to download before shuffling them. The large the buffer, "
"the closer it is to real offline shuffling."
)
},
)
streaming: bool = field(
default=True,
metadata={"help": "Whether to use streaming mode to load and pre-process the data."},
)
@dataclass
class DataCollatorSpeechSeq2SeqWithPadding:
"""
Data collator that will dynamically pad the inputs received.
Args:
processor ([`WhisperProcessor`])
The processor used for processing the data.
decoder_start_token_id (`int`)
The begin-of-sentence of the decoder.
"""
processor: Any
decoder_start_token_id: int
def __call__(self, features: List[Dict[str, Union[List[int], torch.Tensor]]]) -> Dict[str, torch.Tensor]:
# split inputs and labels since they have to be of different lengths and need
# different padding methods
model_input_name = self.processor.model_input_names[0]
input_features = [{model_input_name: feature[model_input_name]} for feature in features]
label_features = [{"input_ids": feature["labels"]} for feature in features]
batch = self.processor.feature_extractor.pad(input_features, return_tensors="pt")
labels_batch = self.processor.tokenizer.pad(label_features, return_tensors="pt")
# replace padding with -100 to ignore loss correctly
labels = labels_batch["input_ids"].masked_fill(labels_batch.attention_mask.ne(1), -100)
# if bos token is appended in previous tokenization step,
# cut bos token here as it's append later anyways
if (labels[:, 0] == self.decoder_start_token_id).all().cpu().item():
labels = labels[:, 1:]
batch["labels"] = labels
return batch
def load_maybe_streaming_dataset(dataset_name, dataset_config_name, split="train", streaming=True, **kwargs):
"""
Utility function to load a dataset in streaming mode. For datasets with multiple splits,
each split is loaded individually and then splits combined by taking alternating examples from
each (interleaving).
"""
if "+" in split:
# load multiple splits separated by the `+` symbol with streaming mode
dataset_splits = [
load_dataset(dataset_name, dataset_config_name, split=split_name, streaming=streaming, **kwargs)
for split_name in split.split("+")
]
# interleave multiple splits to form one dataset
interleaved_dataset = interleave_datasets(dataset_splits)
return interleaved_dataset
else:
# load a single split *with* streaming mode
dataset = load_dataset(dataset_name, dataset_config_name, split=split, streaming=streaming, **kwargs)
return dataset
def main():
# 1. Parse input arguments
# 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, Seq2SeqTrainingArguments))
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_speech_recognition_seq2seq_streaming", model_args, data_args)
# 2. 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)],
)
log_level = training_args.get_process_log_level()
logger.setLevel(log_level)
datasets.utils.logging.set_verbosity(log_level)
transformers.utils.logging.set_verbosity(log_level)
transformers.utils.logging.enable_default_handler()
transformers.utils.logging.enable_explicit_format()
logger.setLevel(logging.INFO if is_main_process(training_args.local_rank) else logging.WARN)
# Log on each process the small summary:
logger.warning(
f"Process rank: {training_args.local_rank}, device: {training_args.device}, n_gpu: {training_args.n_gpu}"
f"distributed training: {bool(training_args.local_rank != -1)}, 16-bits training: {training_args.fp16}"
)
logger.info(f"Training/evaluation parameters {training_args}")
# Set the verbosity to info of the Transformers logger (on main process only):
if is_main_process(training_args.local_rank):
transformers.utils.logging.set_verbosity_info()
logger.info("Training/evaluation parameters %s", training_args)
# 3. 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."
)
# Set seed before initializing model.
set_seed(training_args.seed)
# 4. Load dataset
raw_datasets = IterableDatasetDict() if data_args.streaming else DatasetDict()
if training_args.do_train:
raw_datasets["train"] = load_maybe_streaming_dataset(
data_args.dataset_name,
data_args.dataset_config_name,
split=data_args.train_split_name,
use_auth_token=True if model_args.use_auth_token else None,
streaming=data_args.streaming,
)
if training_args.do_eval:
raw_datasets["eval"] = load_maybe_streaming_dataset(
data_args.dataset_name,
data_args.dataset_config_name,
split=data_args.eval_split_name,
use_auth_token=True if model_args.use_auth_token else None,
streaming=data_args.streaming,
)
raw_datasets_features = list(next(iter(raw_datasets.values())).features.keys())
if data_args.audio_column_name not in raw_datasets_features:
raise ValueError(
f"--audio_column_name '{data_args.audio_column_name}' not found in dataset '{data_args.dataset_name}'. "
"Make sure to set `--audio_column_name` to the correct audio column - one of "
f"{', '.join(raw_datasets_features)}."
)
if data_args.text_column_name not in raw_datasets_features:
raise ValueError(
f"--text_column_name {data_args.text_column_name} not found in dataset '{data_args.dataset_name}'. "
"Make sure to set `--text_column_name` to the correct text column - one of "
f"{', '.join(raw_datasets_features)}."
)
# 5. Load pretrained model, tokenizer, and feature extractor
#
# Distributed training:
# The .from_pretrained methods guarantee that only one local process can concurrently
config = AutoConfig.from_pretrained(
model_args.config_name if model_args.config_name else model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
revision=model_args.model_revision,
use_auth_token=True if model_args.use_auth_token else None,
)
config.update({"forced_decoder_ids": model_args.forced_decoder_ids, "suppress_tokens": model_args.suppress_tokens})
if training_args.gradient_checkpointing:
config.update({"use_cache": False})
feature_extractor = AutoFeatureExtractor.from_pretrained(
model_args.feature_extractor_name if model_args.feature_extractor_name else model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
revision=model_args.model_revision,
use_auth_token=True if model_args.use_auth_token else None,
)
tokenizer = AutoTokenizer.from_pretrained(
model_args.tokenizer_name if model_args.tokenizer_name else model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
use_fast=model_args.use_fast_tokenizer,
revision=model_args.model_revision,
use_auth_token=True if model_args.use_auth_token else None,
)
model = AutoModelForSpeechSeq2Seq.from_pretrained(
model_args.model_name_or_path,
config=config,
cache_dir=model_args.cache_dir,
revision=model_args.model_revision,
use_auth_token=True if model_args.use_auth_token else None,
)
if model.config.decoder_start_token_id is None:
raise ValueError("Make sure that `config.decoder_start_token_id` is correctly defined")
if model_args.freeze_feature_encoder:
model.freeze_feature_encoder()
if model_args.freeze_encoder:
model.freeze_encoder()
if data_args.language is not None:
# We only need to set the task id when the language is specified (i.e. in a multilingual setting)
tokenizer.set_prefix_tokens(language=data_args.language, task=data_args.task)
# 6. Resample speech dataset if necessary
dataset_sampling_rate = next(iter(raw_datasets.values())).features[data_args.audio_column_name].sampling_rate
if dataset_sampling_rate != feature_extractor.sampling_rate:
raw_datasets = raw_datasets.cast_column(
data_args.audio_column_name, datasets.features.Audio(sampling_rate=feature_extractor.sampling_rate)
)
# 7. Preprocessing the datasets.
# We need to read the audio files as arrays and tokenize the targets.
max_input_length = data_args.max_duration_in_seconds * feature_extractor.sampling_rate
min_input_length = data_args.min_duration_in_seconds * feature_extractor.sampling_rate
audio_column_name = data_args.audio_column_name
text_column_name = data_args.text_column_name
model_input_name = feature_extractor.model_input_names[0]
do_lower_case = data_args.do_lower_case
do_remove_punctuation = data_args.do_remove_punctuation
normalizer = BasicTextNormalizer() # 'official' text normalizer from OpenAI
if data_args.max_train_samples is not None:
raw_datasets["train"] = (
raw_datasets["train"].take(data_args.max_train_samples)
if data_args.streaming
else raw_datasets["train"].select(range(data_args.max_train_samples))
)
if data_args.max_eval_samples is not None:
raw_datasets["eval"] = (
raw_datasets["eval"].take(data_args.max_eval_samples)
if data_args.streaming
else raw_datasets["eval"].select(range(data_args.max_eval_samples))
)
def prepare_dataset(batch):
# process audio
sample = batch[audio_column_name]
inputs = feature_extractor(sample["array"], sampling_rate=sample["sampling_rate"])
# process audio length
batch[model_input_name] = inputs.get(model_input_name)[0]
batch["input_length"] = len(sample["array"])
# process targets
input_str = batch[text_column_name].lower() if do_lower_case else batch[text_column_name]
if do_remove_punctuation:
input_str = normalizer(input_str).strip()
batch["labels"] = tokenizer(input_str).input_ids
return batch
with training_args.main_process_first(desc="dataset map pre-processing"):
vectorized_datasets = raw_datasets.map(
prepare_dataset,
remove_columns=raw_datasets_features,
).with_format("torch")
if training_args.do_train and data_args.streaming:
# manually shuffle if streaming (done by the trainer for non-streaming)
vectorized_datasets["train"] = vectorized_datasets["train"].shuffle(
buffer_size=data_args.shuffle_buffer_size,
seed=training_args.seed,
)
# filter training data that is shorter than min_input_length or longer than
# max_input_length
def is_audio_in_length_range(length):
return min_input_length < length < max_input_length
vectorized_datasets["train"] = vectorized_datasets["train"].filter(
is_audio_in_length_range,
input_columns=["input_length"],
)
# 8. Load Metric
metric = evaluate.load("wer")
do_normalize_eval = data_args.do_normalize_eval
def compute_metrics(pred):
pred_ids = pred.predictions
pred.label_ids[pred.label_ids == -100] = tokenizer.pad_token_id
pred_str = tokenizer.batch_decode(pred_ids, skip_special_tokens=True)
# we do not want to group tokens when computing the metrics
label_str = tokenizer.batch_decode(pred.label_ids, skip_special_tokens=True)
if do_normalize_eval:
pred_str = [normalizer(pred) for pred in pred_str]
label_str = [normalizer(label) for label in label_str]
wer = 100 * metric.compute(predictions=pred_str, references=label_str)
return {"wer": wer}
# 9. Create a single speech processor
if is_main_process(training_args.local_rank):
# save feature extractor, tokenizer and config
feature_extractor.save_pretrained(training_args.output_dir)
tokenizer.save_pretrained(training_args.output_dir)
config.save_pretrained(training_args.output_dir)
processor = AutoProcessor.from_pretrained(training_args.output_dir)
# 10. Define data collator
data_collator = DataCollatorSpeechSeq2SeqWithPadding(
processor=processor,
decoder_start_token_id=model.config.decoder_start_token_id,
)
# 11. Configure Trainer
# Trainer callback to reinitialise and reshuffle the streamable datasets at the beginning of each epoch
# Only required for streaming: Trainer automatically shuffles non-streaming datasets
class ShuffleCallback(TrainerCallback):
def on_epoch_begin(self, args, state, control, train_dataloader, **kwargs):
if isinstance(train_dataloader.dataset, IterableDatasetShard):
pass # set_epoch() is handled by the Trainer
elif isinstance(train_dataloader.dataset, IterableDataset):
train_dataloader.dataset.set_epoch(train_dataloader.dataset._epoch + 1)
# Initialize Trainer
trainer = Seq2SeqTrainer(
model=model,
args=training_args,
train_dataset=vectorized_datasets["train"] if training_args.do_train else None,
eval_dataset=vectorized_datasets["eval"] if training_args.do_eval else None,
tokenizer=feature_extractor,
data_collator=data_collator,
compute_metrics=compute_metrics if training_args.predict_with_generate else None,
callbacks=[ShuffleCallback()] if data_args.streaming else None,
)
# 12. Training
if training_args.do_train:
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
train_result = trainer.train(resume_from_checkpoint=checkpoint)
trainer.save_model() # Saves the feature extractor too for easy upload
metrics = train_result.metrics
if data_args.max_train_samples:
metrics["train_samples"] = data_args.max_train_samples
trainer.log_metrics("train", metrics)
trainer.save_metrics("train", metrics)
trainer.save_state()
# 13. Evaluation
results = {}
if training_args.do_eval:
logger.info("*** Evaluate ***")
metrics = trainer.evaluate(
metric_key_prefix="eval",
max_length=training_args.generation_max_length,
num_beams=training_args.generation_num_beams,
)
if data_args.max_eval_samples:
metrics["eval_samples"] = data_args.max_eval_samples
trainer.log_metrics("eval", metrics)
trainer.save_metrics("eval", metrics)
# 14. Write Training Stats
kwargs = {
"finetuned_from": model_args.model_name_or_path,
"tasks": "automatic-speech-recognition",
"tags": "whisper-event",
}
if data_args.dataset_name is not None:
kwargs["dataset_tags"] = data_args.dataset_name
if data_args.dataset_config_name is not None:
kwargs["dataset"] = f"{data_args.dataset_name} {data_args.dataset_config_name}"
else:
kwargs["dataset"] = data_args.dataset_name
if "common_voice" in data_args.dataset_name:
kwargs["language"] = data_args.dataset_config_name[:2]
if model_args.model_index_name is not None:
kwargs["model_name"] = model_args.model_index_name
if training_args.push_to_hub:
trainer.push_to_hub(**kwargs)
else:
trainer.create_model_card(**kwargs)
return results
if __name__ == "__main__":
main()