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Add the XTREME-S fine-tuning example (#15985)

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<!---
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.
-->
# XTREME-S benchmark examples
*Maintainers: [Anton Lozhkov](https://github.com/anton-l) and [Patrick von Platen](https://github.com/patrickvonplaten)*
The Cross-lingual TRansfer Evaluation of Multilingual Encoders for Speech (XTREME-S) benchmark is a benchmark designed to evaluate speech representations across languages, tasks, domains and data regimes. It covers XX typologically diverse languages and seven downstream tasks grouped in four families: speech recognition, translation, classification and retrieval.
XTREME-S covers speech recognition with BABEL, Multilingual LibriSpeech (MLS) and VoxPopuli, speech translation with CoVoST-2, speech classification with LangID (FLoRes) and intent classification (MInds-14) and finally speech retrieval with speech-speech translation data mining (bi-speech retrieval). Each of the tasks covers a subset of the 40 languages included in XTREME-S (shown here with their ISO 639-1 codes): ar, as, ca, cs, cy, da, de, en, en, en, en, es, et, fa, fi, fr, hr, hu, id, it, ja, ka, ko, lo, lt, lv, mn, nl, pl, pt, ro, ru, sk, sl, sv, sw, ta, tl, tr and zh.
Paper: `<TODO>`
Dataset: [https://huggingface.co/datasets/google/xtreme_s](https://huggingface.co/datasets/google/xtreme_s)
## Fine-tuning for the XTREME-S tasks
Based on the [`run_xtreme_s.py`](https://github.com/huggingface/transformers/blob/master/examples/research_projects/xtreme-s/run_xtreme_s.py) script.
This script can fine-tune any of the pretrained speech models on the [hub](https://huggingface.co/models?pipeline_tag=automatic-speech-recognition) on the [XTREME-S dataset](https://huggingface.co/datasets/google/xtreme_s) tasks.
XTREME-S is made up of 7 different task-specific subsets. Here is how to run the script on each of them:
```bash
export TASK_NAME=mls.all
python run_xtreme_s.py \
--model_name_or_path="facebook/wav2vec2-xls-r-300m" \
--dataset_name="google/xtreme_s" \
--dataset_config_name="${TASK_NAME}" \
--eval_split_name="validation" \
--output_dir="xtreme_s_xlsr_${TASK_NAME}" \
--num_train_epochs=100 \
--per_device_train_batch_size=32 \
--learning_rate="3e-4" \
--target_column_name="transcription" \
--save_steps=500 \
--eval_steps=500 \
--freeze_feature_encoder \
--gradient_checkpointing \
--fp16 \
--group_by_length \
--do_train \
--do_eval \
--push_to_hub
```
where `TASK_NAME` can be one of: `mls.all, voxpopuli, covost2.all, fleurs.all, minds14.all`.
We get the following results on the test set of the benchmark's datasets.
The corresponding training commands for each dataset are given in the sections below:
| Task | Dataset | Result | Fine-tuned model & logs | Training time | GPUs |
|-----------------------|-----------|-----------------------|--------------------------------------------------------------------|---------------|--------|
| Speech Recognition | MLS | 30.33 WER | [here](https://huggingface.co/anton-l/xtreme_s_xlsr_300m_mls/) | 18:47:25 | 8xV100 |
| Speech Recognition | VoxPopuli | - | - | - | - |
| Speech Recognition | FLEURS | - | - | - | - |
| Speech Translation | CoVoST-2 | - | - | - | - |
| Speech Classification | Minds-14 | 94.74 F1 / 94.70 Acc. | [here](https://huggingface.co/anton-l/xtreme_s_xlsr_300m_minds14/) | 04:46:40 | 2xA100 |
| Speech Classification | FLEURS | - | - | - | - |
| Speech Retrieval | FLEURS | - | - | - | - |
### Speech Recognition with MLS
The following command shows how to fine-tune the [XLS-R](https://huggingface.co/docs/transformers/master/model_doc/xls_r) model on [XTREME-S MLS](https://huggingface.co/datasets/google/xtreme_s#multilingual-librispeech-mls) using 8 GPUs in half-precision.
```bash
python -m torch.distributed.launch \
--nproc_per_node=8 \
run_xtreme_s.py \
--model_name_or_path="facebook/wav2vec2-xls-r-300m" \
--dataset_name="google/xtreme_s" \
--dataset_config_name="mls.all" \
--eval_split_name="test" \
--output_dir="xtreme_s_xlsr_300m_mls" \
--overwrite_output_dir \
--num_train_epochs=100 \
--per_device_train_batch_size=4 \
--per_device_eval_batch_size=1 \
--gradient_accumulation_steps=2 \
--learning_rate="3e-4" \
--warmup_steps=3000 \
--evaluation_strategy="steps" \
--target_column_name="transcription" \
--max_duration_in_seconds=20 \
--save_steps=500 \
--eval_steps=500 \
--logging_steps=1 \
--layerdrop=0.0 \
--mask_time_prob=0.3 \
--mask_time_length=10 \
--mask_feature_prob=0.1 \
--mask_feature_length=64 \
--freeze_feature_encoder \
--gradient_checkpointing \
--fp16 \
--group_by_length \
--do_train \
--do_eval \
--metric_for_best_model="wer" \
--greater_is_better=False \
--load_best_model_at_end \
--push_to_hub
```
On 8 V100 GPUs, this script should run in ~19 hours and yield a cross-entropy loss of **0.6215** and word error rate of **30.33**
### Speech Classification with Minds-14
The following command shows how to fine-tune the [XLS-R](https://huggingface.co/docs/transformers/master/model_doc/xls_r) model on [XTREME-S MLS](https://huggingface.co/datasets/google/xtreme_s#intent-classification---minds-14) using 2 GPUs in half-precision.
```bash
python -m torch.distributed.launch \
--nproc_per_node=2 \
run_xtreme_s.py \
--model_name_or_path="facebook/wav2vec2-xls-r-300m" \
--dataset_name="google/xtreme_s" \
--dataset_config_name="minds14.all" \
--eval_split_name="test" \
--output_dir="xtreme_s_xlsr_300m_minds14" \
--overwrite_output_dir \
--num_train_epochs=50 \
--per_device_train_batch_size=32 \
--per_device_eval_batch_size=8 \
--gradient_accumulation_steps=1 \
--learning_rate="3e-4" \
--warmup_steps=1500 \
--evaluation_strategy="steps" \
--target_column_name="intent_class" \
--max_duration_in_seconds=30 \
--save_steps=200 \
--eval_steps=200 \
--logging_steps=1 \
--layerdrop=0.0 \
--mask_time_prob=0.3 \
--mask_time_length=10 \
--mask_feature_prob=0.1 \
--mask_feature_length=64 \
--freeze_feature_encoder \
--gradient_checkpointing \
--fp16 \
--group_by_length \
--do_train \
--do_eval \
--metric_for_best_model="f1" \
--greater_is_better=True \
--load_best_model_at_end \
--push_to_hub
```
On 2 A100 GPUs, this script should run in ~5 hours and yield a cross-entropy loss of **0.2890** and F1 score of **94.74**

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datasets >= 1.18.0
torch >= 1.5
torchaudio
librosa
jiwer

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#!/usr/bin/env python
# coding=utf-8
# Copyright 2022 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
""" Fine-tuning a 🤗 Transformers pretrained speech model on the XTREME-S benchmark tasks"""
import functools
import json
import logging
import os
import re
import sys
from dataclasses import dataclass, field
from typing import Dict, List, Optional, Union
import datasets
import numpy as np
import torch
from datasets import DatasetDict, load_dataset, load_metric
import transformers
from transformers import (
AutoConfig,
AutoFeatureExtractor,
AutoModelForAudioClassification,
AutoModelForCTC,
AutoModelForSpeechSeq2Seq,
AutoProcessor,
AutoTokenizer,
HfArgumentParser,
Seq2SeqTrainer,
Seq2SeqTrainingArguments,
Trainer,
set_seed,
)
from transformers.trainer_utils import get_last_checkpoint, is_main_process
from transformers.utils import check_min_version
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("4.18.0.dev0")
require_version("datasets>=1.18.0", "To fix: pip install -r examples/pytorch/speech-recognition/requirements.txt")
logger = logging.getLogger(__name__)
def list_field(default=None, metadata=None):
return field(default_factory=lambda: default, metadata=metadata)
@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"}
)
tokenizer_name_or_path: Optional[str] = field(
default=None,
metadata={"help": "Path to pretrained tokenizer or tokenizer identifier from huggingface.co/models"},
)
cache_dir: Optional[str] = field(
default=None,
metadata={
"help": "Where do you want to store the pretrained models and datasets downloaded from " "huggingface.co"
},
)
freeze_feature_encoder: bool = field(
default=True, metadata={"help": "Whether to freeze the feature encoder layers of the model."}
)
attention_dropout: float = field(
default=0.0, metadata={"help": "The dropout ratio for the attention probabilities."}
)
activation_dropout: float = field(
default=0.0, metadata={"help": "The dropout ratio for activations inside the fully connected layer."}
)
feat_proj_dropout: float = field(default=0.0, metadata={"help": "The dropout ratio for the projected features."})
hidden_dropout: float = field(
default=0.0,
metadata={
"help": "The dropout probability for all fully connected layers in the embeddings, encoder, and pooler."
},
)
final_dropout: float = field(
default=0.0,
metadata={"help": "The dropout probability for the final projection layer."},
)
mask_time_prob: float = field(
default=0.05,
metadata={
"help": "Probability of each feature vector along the time axis to be chosen as the start of the vector"
"span to be masked. Approximately ``mask_time_prob * sequence_length // mask_time_length`` feature"
"vectors will be masked along the time axis."
},
)
mask_time_length: int = field(
default=10,
metadata={"help": "Length of vector span to mask along the time axis."},
)
mask_feature_prob: float = field(
default=0.0,
metadata={
"help": "Probability of each feature vector along the feature axis to be chosen as the start of the vector"
"span to be masked. Approximately ``mask_feature_prob * sequence_length // mask_feature_length`` feature bins will be masked along the time axis."
},
)
mask_feature_length: int = field(
default=10,
metadata={"help": "Length of vector span to mask along the feature axis."},
)
layerdrop: float = field(default=0.0, metadata={"help": "The LayerDrop probability."})
ctc_loss_reduction: Optional[str] = field(
default="mean", metadata={"help": "The way the ctc loss should be reduced. Should be one of 'mean' or 'sum'."}
)
@dataclass
class DataTrainingArguments:
"""
Arguments pertaining to what data we are going to input our model for training and eval.
Using `HfArgumentParser` we can turn this class
into argparse arguments to be able to specify them on
the command line.
"""
dataset_name: str = field(
default="xtreme_s",
metadata={"help": "The name of the dataset to use (via the datasets library). Defaults to 'xtreme_s'"},
)
dataset_config_name: str = field(
default=None, metadata={"help": "The configuration name of the dataset to use (via the datasets library)."}
)
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="validation",
metadata={
"help": "The name of the evaluation data set split to use (via the datasets library). "
"Defaults to 'validation'"
},
)
audio_column_name: str = field(
default="audio",
metadata={"help": "The name of the dataset column containing the audio data. Defaults to 'audio'"},
)
target_column_name: str = field(
default="transcription",
metadata={
"help": "The name of the dataset column containing the target data "
"(transcription/translation/label). Defaults to 'transcription'"
},
)
overwrite_cache: bool = field(
default=False, metadata={"help": "Overwrite the cached preprocessed datasets or not."}
)
preprocessing_num_workers: Optional[int] = field(
default=None,
metadata={"help": "The number of processes to use for the preprocessing."},
)
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 validation examples to this "
"value if set."
},
)
chars_to_ignore: Optional[List[str]] = list_field(
default=', ? . ! - ; : " “ % ‘ ” <20>'.split(" "),
metadata={"help": "A list of characters to remove from the transcripts."},
)
max_duration_in_seconds: float = field(
default=30.0,
metadata={
"help": "Filter 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"}
)
preprocessing_only: bool = field(
default=False,
metadata={
"help": "Whether to only do data preprocessing and skip training. "
"This is especially useful when data preprocessing errors out in distributed training due to timeout. "
"In this case, one should run the preprocessing in a non-distributed setup with `preprocessing_only=True` "
"so that the cached datasets can consequently be loaded in distributed training"
},
)
use_auth_token: bool = field(
default=False,
metadata={
"help": "If :obj:`True`, will use the token generated when running"
":obj:`transformers-cli login` as HTTP bearer authorization for remote files."
},
)
unk_token: str = field(
default="[UNK]",
metadata={"help": "The unk token for the tokenizer"},
)
pad_token: str = field(
default="[PAD]",
metadata={"help": "The padding token for the tokenizer"},
)
word_delimiter_token: str = field(
default="|",
metadata={"help": "The word delimiter token for the tokenizer"},
)
phoneme_language: Optional[str] = field(
default=None,
metadata={
"help": "The target language that should be used be"
" passed to the tokenizer for tokenization. Note that"
" this is only relevant if the model classifies the"
" input audio to a sequence of phoneme sequences."
},
)
@dataclass
class SpeechDataCollatorWithPadding:
processor: AutoProcessor
decoder_start_token_id: Optional[int] = None
padding: Union[bool, str] = "longest"
pad_labels: Optional[int] = True
pad_to_multiple_of: Optional[int] = None
pad_to_multiple_of_labels: Optional[int] = None
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 lenghts and need
# different padding methods
input_features = [{"input_values": feature["input_values"]} for feature in features]
batch = self.processor.pad(
input_features,
padding=self.padding,
pad_to_multiple_of=self.pad_to_multiple_of,
return_tensors="pt",
)
if self.pad_labels:
label_features = [{"input_ids": feature["labels"]} for feature in features]
with self.processor.as_target_processor():
labels_batch = self.processor.pad(
label_features,
padding=self.padding,
pad_to_multiple_of=self.pad_to_multiple_of_labels,
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 (
self.decoder_start_token_id is not None
and (labels[:, 0] == self.decoder_start_token_id).all().cpu().item()
):
labels = labels[:, 1:]
batch["labels"] = labels
else:
batch["labels"] = torch.tensor([feature["labels"] for feature in features])
return batch
def create_vocabulary_from_data(
datasets: DatasetDict,
word_delimiter_token: Optional[str] = None,
unk_token: Optional[str] = None,
pad_token: Optional[str] = None,
):
# Given training and test labels create vocabulary
def extract_all_chars(batch):
all_text = " ".join(batch["target_text"])
vocab = list(set(all_text))
return {"vocab": [vocab], "all_text": [all_text]}
vocabs = datasets.map(
extract_all_chars,
batched=True,
batch_size=-1,
keep_in_memory=True,
remove_columns=datasets["train"].column_names,
)
# take union of all unique characters in each dataset
vocab_set = functools.reduce(
lambda vocab_1, vocab_2: set(vocab_1["vocab"][0]) | set(vocab_2["vocab"][0]), vocabs.values()
)
vocab_dict = {v: k for k, v in enumerate(sorted(list(vocab_set)))}
# replace white space with delimiter token
if word_delimiter_token is not None:
vocab_dict[word_delimiter_token] = vocab_dict[" "]
del vocab_dict[" "]
# add unk and pad token
if unk_token is not None:
vocab_dict[unk_token] = len(vocab_dict)
if pad_token is not None:
vocab_dict[pad_token] = len(vocab_dict)
return vocab_dict
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, 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()
# Detecting 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:
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 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}"
)
# 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)
# Set seed before initializing model.
set_seed(training_args.seed)
# 1. First, let's load the dataset
raw_datasets = DatasetDict()
if data_args.dataset_config_name is None:
raise ValueError(
"Set --dataset_config_name should be set to '<xtreme_s_subset>.<language(s)>' "
"(e.g. 'mls.pl', 'covost2.en.tr', 'minds14.fr-FR') "
"or '<xtreme_s_subset>.all' for multi-lingual fine-tuning."
)
task_name = data_args.dataset_config_name.split(".")[0]
target_column_name = data_args.target_column_name
# here we differentiate between tasks with text as the target and classification tasks
is_text_target = target_column_name in ("transcription", "translation")
if training_args.do_train:
raw_datasets["train"] = load_dataset(
data_args.dataset_name,
data_args.dataset_config_name,
split=data_args.train_split_name,
use_auth_token=data_args.use_auth_token,
cache_dir=model_args.cache_dir,
)
if data_args.audio_column_name not in raw_datasets["train"].column_names:
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['train'].column_names)}."
)
if data_args.target_column_name not in raw_datasets["train"].column_names:
raise ValueError(
f"--target_column_name {data_args.target_column_name} not found in dataset '{data_args.dataset_name}'. "
"Make sure to set `--target_column_name` to the correct text 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 not is_text_target:
label_list = raw_datasets["train"].features[data_args.target_column_name].names
num_labels = len(label_list)
if training_args.do_eval:
raw_datasets["eval"] = load_dataset(
data_args.dataset_name,
data_args.dataset_config_name,
split=data_args.eval_split_name,
use_auth_token=data_args.use_auth_token,
cache_dir=model_args.cache_dir,
)
if data_args.max_eval_samples is not None:
raw_datasets["eval"] = raw_datasets["eval"].select(range(data_args.max_eval_samples))
# 2. We remove some special characters from the datasets
# that make training complicated and do not help in transcribing the speech
# E.g. characters, such as `,` and `.` do not really have an acoustic characteristic
# that could be easily picked up by the model
chars_to_ignore_regex = (
f'[{"".join(data_args.chars_to_ignore)}]' if data_args.chars_to_ignore is not None else None
)
def remove_special_characters(batch):
if chars_to_ignore_regex is not None:
batch["target_text"] = re.sub(chars_to_ignore_regex, "", batch[target_column_name]).lower() + " "
else:
batch["target_text"] = batch[target_column_name].lower() + " "
return batch
if is_text_target:
with training_args.main_process_first(desc="dataset map special characters removal"):
raw_datasets = raw_datasets.map(
remove_special_characters,
remove_columns=[target_column_name],
desc="remove special characters from datasets",
)
# save special tokens for tokenizer
word_delimiter_token = data_args.word_delimiter_token
unk_token = data_args.unk_token
pad_token = data_args.pad_token
# 3. Next, let's load the config as we might need it to create
# the tokenizer
config = AutoConfig.from_pretrained(
model_args.model_name_or_path, cache_dir=model_args.cache_dir, use_auth_token=data_args.use_auth_token
)
if is_text_target:
# 4. (Optional, for ASR and translation) If no tokenizer file is defined,
# we create the vocabulary of the model by extracting all unique characters from
# the training and evaluation datasets
# We need to make sure that only first rank saves vocabulary
# make sure all processes wait until vocab is created
tokenizer_name_or_path = model_args.tokenizer_name_or_path
tokenizer_kwargs = {}
if tokenizer_name_or_path is None:
# save vocab in training output dir
tokenizer_name_or_path = training_args.output_dir
vocab_file = os.path.join(tokenizer_name_or_path, "vocab.json")
with training_args.main_process_first():
if training_args.overwrite_output_dir and os.path.isfile(vocab_file):
os.remove(vocab_file)
with training_args.main_process_first(desc="dataset map vocabulary creation"):
if not os.path.isfile(vocab_file):
os.makedirs(tokenizer_name_or_path, exist_ok=True)
vocab_dict = create_vocabulary_from_data(
raw_datasets,
word_delimiter_token=word_delimiter_token,
unk_token=unk_token,
pad_token=pad_token,
)
# save vocab dict to be loaded into tokenizer
with open(vocab_file, "w") as file:
json.dump(vocab_dict, file)
# if tokenizer has just been created
# it is defined by `tokenizer_class` if present in config else by `model_type`
if not config.is_encoder_decoder:
tokenizer_kwargs = {
"config": config if config.tokenizer_class is not None else None,
"tokenizer_type": config.model_type if config.tokenizer_class is None else None,
"unk_token": unk_token,
"pad_token": pad_token,
"word_delimiter_token": word_delimiter_token,
}
else:
tokenizer_kwargs = {}
# 5. Now we can instantiate the feature extractor, tokenizer and model
# Note for distributed training, the .from_pretrained methods guarantee that only
# one local process can concurrently download model & vocab.
# load feature_extractor and tokenizer
if is_text_target:
tokenizer = AutoTokenizer.from_pretrained(
tokenizer_name_or_path,
use_auth_token=data_args.use_auth_token,
**tokenizer_kwargs,
)
feature_extractor = AutoFeatureExtractor.from_pretrained(
model_args.model_name_or_path, cache_dir=model_args.cache_dir, use_auth_token=data_args.use_auth_token
)
# adapt config
config.update(
{
"feat_proj_dropout": model_args.feat_proj_dropout,
"attention_dropout": model_args.attention_dropout,
"hidden_dropout": model_args.hidden_dropout,
"final_dropout": model_args.final_dropout,
"mask_time_prob": model_args.mask_time_prob,
"mask_time_length": model_args.mask_time_length,
"mask_feature_prob": model_args.mask_feature_prob,
"mask_feature_length": model_args.mask_feature_length,
"gradient_checkpointing": training_args.gradient_checkpointing,
"layerdrop": model_args.layerdrop,
"ctc_loss_reduction": model_args.ctc_loss_reduction,
"activation_dropout": model_args.activation_dropout,
}
)
if training_args.do_train:
if is_text_target:
config.pad_token_id = tokenizer.pad_token_id
config.vocab_size = len(tokenizer)
else:
label_to_id = {v: i for i, v in enumerate(label_list)}
config.label2id = label_to_id
config.id2label = {id: label for label, id in label_to_id.items()}
config.num_labels = num_labels
# create model
if target_column_name == "transcription":
model = AutoModelForCTC.from_pretrained(
model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
config=config,
use_auth_token=data_args.use_auth_token,
)
elif config.is_encoder_decoder:
model = AutoModelForSpeechSeq2Seq.from_pretrained(
model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
config=config,
use_auth_token=data_args.use_auth_token,
)
if model.config.decoder_start_token_id is None:
raise ValueError("Make sure that `config.decoder_start_token_id` is correctly defined")
else:
model = AutoModelForAudioClassification.from_pretrained(
model_args.model_name_or_path,
cache_dir=model_args.cache_dir,
config=config,
use_auth_token=data_args.use_auth_token,
)
# freeze encoder
if model_args.freeze_feature_encoder:
model.freeze_feature_encoder()
# 6. 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`
# make sure that dataset decodes audio with correct sampling rate
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)
)
# derive max & min input length for sample rate & max duration
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
num_workers = data_args.preprocessing_num_workers
# `phoneme_language` is only relevant if the model is fine-tuned on phoneme classification
phoneme_language = data_args.phoneme_language
# Preprocessing the datasets.
# We need to read the audio files as arrays and tokenize the targets.
def prepare_dataset(batch):
# load audio
sample = batch[audio_column_name]
inputs = feature_extractor(sample["array"], sampling_rate=sample["sampling_rate"])
batch["input_values"] = inputs.input_values[0]
batch["length"] = len(batch["input_values"])
# encode targets
additional_kwargs = {}
if phoneme_language is not None:
additional_kwargs["phonemizer_lang"] = phoneme_language
if is_text_target:
batch["labels"] = tokenizer(batch["target_text"], **additional_kwargs).input_ids
else:
batch["labels"] = batch[data_args.target_column_name]
return batch
with training_args.main_process_first(desc="dataset map preprocessing"):
vectorized_datasets = raw_datasets.map(
prepare_dataset,
remove_columns=next(iter(raw_datasets.values())).column_names,
num_proc=num_workers,
desc="preprocess datasets",
)
if training_args.do_train:
def is_audio_in_length_range(length):
return length > min_input_length and length < max_input_length
# filter data that is shorter than min_input_length
vectorized_datasets["train"] = vectorized_datasets["train"].filter(
is_audio_in_length_range,
num_proc=num_workers,
input_columns=["length"],
)
# 7. Next, we can prepare for the training step.
# Let's use the appropriate XTREME-S evaluation metric,
# instantiate a data collator and the trainer
# Define evaluation metrics during training, *i.e.* word error rate, character error rate
eval_metric = load_metric("xtreme_s", task_name)
# 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:
logger.info(f"Data preprocessing finished. Files cached at {vectorized_datasets.cache_files}")
return
def compute_asr_metric(pred):
pred_logits = pred.predictions
pred_ids = np.argmax(pred_logits, axis=-1)
pred.label_ids[pred.label_ids == -100] = tokenizer.pad_token_id
pred_str = tokenizer.batch_decode(pred_ids)
# we do not want to group tokens when computing the metrics
label_str = tokenizer.batch_decode(pred.label_ids, group_tokens=False)
metric = eval_metric.compute(predictions=pred_str, references=label_str)
return metric
def compute_classification_metric(pred):
pred_ids = np.argmax(pred.predictions, axis=1)
metric = eval_metric.compute(predictions=pred_ids, references=pred.label_ids)
return metric
# Now save everything to be able to create a single processor later
if is_main_process(training_args.local_rank):
# save feature extractor, tokenizer and config
feature_extractor.save_pretrained(training_args.output_dir)
if is_text_target:
tokenizer.save_pretrained(training_args.output_dir)
config.save_pretrained(training_args.output_dir)
# wait until configs are saved in the main process before loading the processor
torch.distributed.barrier()
if is_text_target:
processor = AutoProcessor.from_pretrained(training_args.output_dir)
else:
processor = AutoFeatureExtractor.from_pretrained(training_args.output_dir)
# Instantiate custom data collator
data_collator = SpeechDataCollatorWithPadding(processor=processor, pad_labels=is_text_target)
# Initialize Trainer
if target_column_name == "translation":
trainer = Seq2SeqTrainer(
model=model,
data_collator=data_collator,
args=training_args,
compute_metrics=compute_asr_metric if training_args.predict_with_generate else None,
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,
)
else:
trainer = Trainer(
model=model,
data_collator=data_collator,
args=training_args,
compute_metrics=compute_asr_metric if is_text_target else compute_classification_metric,
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,
)
# 8. Finally, we can start training
# Training
if training_args.do_train:
# use last checkpoint if exist
if last_checkpoint is not None:
checkpoint = last_checkpoint
elif os.path.isdir(model_args.model_name_or_path):
checkpoint = model_args.model_name_or_path
else:
checkpoint = None
train_result = trainer.train(resume_from_checkpoint=checkpoint)
trainer.save_model()
metrics = train_result.metrics
max_train_samples = (
data_args.max_train_samples
if data_args.max_train_samples is not None
else len(vectorized_datasets["train"])
)
metrics["train_samples"] = min(max_train_samples, len(vectorized_datasets["train"]))
trainer.log_metrics("train", metrics)
trainer.save_metrics("train", metrics)
trainer.save_state()
# Evaluation
results = {}
if training_args.do_eval:
logger.info("*** Evaluate ***")
metrics = trainer.evaluate()
max_eval_samples = (
data_args.max_eval_samples if data_args.max_eval_samples is not None else len(vectorized_datasets["eval"])
)
metrics["eval_samples"] = min(max_eval_samples, len(vectorized_datasets["eval"]))
trainer.log_metrics("eval", metrics)
trainer.save_metrics("eval", metrics)
# Write model card and (optionally) push to hub
config_name = data_args.dataset_config_name if data_args.dataset_config_name is not None else "na"
kwargs = {
"finetuned_from": model_args.model_name_or_path,
"tasks": "speech-recognition",
"tags": ["automatic-speech-recognition", data_args.dataset_name],
"dataset_args": f"Config: {config_name}, Training split: {data_args.train_split_name}, Eval split: {data_args.eval_split_name}",
"dataset": f"{data_args.dataset_name.upper()} - {config_name.upper()}",
}
if "common_voice" in data_args.dataset_name:
kwargs["language"] = config_name
if training_args.push_to_hub:
trainer.push_to_hub(**kwargs)
else:
trainer.create_model_card(**kwargs)
return results
if __name__ == "__main__":
main()