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Model: mikr/whisper-large-v3-czech-cv13 Source: Original Platform
This commit is contained in:
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67
README.md
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README.md
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---
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license: apache-2.0
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base_model: openai/whisper-large-v3
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tags:
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- generated_from_trainer
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metrics:
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- wer
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model-index:
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- name: openai/whisper-large-v3
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results: []
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---
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<!-- This model card has been generated automatically according to the information the Trainer had access to. You
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should probably proofread and complete it, then remove this comment. -->
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# openai/whisper-large-v3
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This model is a fine-tuned version of [openai/whisper-large-v3](https://huggingface.co/openai/whisper-large-v3) on the None dataset.
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It achieves the following results on the evaluation set:
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- Loss: 0.1283
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- Wer: 0.0789
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## Model description
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More information needed
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## Intended uses & limitations
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More information needed
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## Training and evaluation data
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More information needed
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## Training procedure
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### Training hyperparameters
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The following hyperparameters were used during training:
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- learning_rate: 1e-05
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- train_batch_size: 62
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- eval_batch_size: 16
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- seed: 42
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- distributed_type: multi-GPU
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- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
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- lr_scheduler_type: linear
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- lr_scheduler_warmup_steps: 500
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- training_steps: 5000
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- mixed_precision_training: Native AMP
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### Training results
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| Training Loss | Epoch | Step | Validation Loss | Wer |
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|:-------------:|:-----:|:----:|:---------------:|:------:|
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| 0.0138 | 2.24 | 1000 | 0.0962 | 0.0863 |
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| 0.004 | 4.48 | 2000 | 0.1117 | 0.0844 |
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| 0.0015 | 6.73 | 3000 | 0.1178 | 0.0807 |
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| 0.0004 | 8.97 | 4000 | 0.1219 | 0.0792 |
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| 0.0002 | 11.21 | 5000 | 0.1283 | 0.0789 |
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### Framework versions
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- Transformers 4.36.0.dev0
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- Pytorch 2.0.0+cu117
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- Datasets 2.14.6
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- Tokenizers 0.14.1
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config.json
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config.json
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{
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"_name_or_path": "openai/whisper-large-v3",
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50001
merges.txt
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3
model.safetensors
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3
model.safetensors
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version https://git-lfs.github.com/spec/v1
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normalizer.json
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14
preprocessor_config.json
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preprocessor_config.json
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{
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|
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"return_attention_mask": false,
|
||||
"sampling_rate": 16000
|
||||
}
|
||||
7
requirements.txt
Normal file
7
requirements.txt
Normal file
@@ -0,0 +1,7 @@
|
||||
datasets >= 1.18.0
|
||||
git+https://github.com/huggingface/transformers
|
||||
torch >= 1.5
|
||||
torchaudio
|
||||
librosa
|
||||
jiwer
|
||||
evaluate
|
||||
39
run-stream.sh
Normal file
39
run-stream.sh
Normal file
@@ -0,0 +1,39 @@
|
||||
deepspeed run_speech_recognition_seq2seq_streaming.py \
|
||||
--deepspeed="ds_config.json" \
|
||||
--model_name_or_path="openai/whisper-large-v3" \
|
||||
--dataset_name="mozilla-foundation/common_voice_13_0" \
|
||||
--dataset_config_name="cs" \
|
||||
--language="czech" \
|
||||
--train_split_name="train+validation" \
|
||||
--eval_split_name="test" \
|
||||
--max_steps="5000" \
|
||||
--output_dir="./" \
|
||||
--per_device_train_batch_size="20" \
|
||||
--per_device_eval_batch_size="16" \
|
||||
--gradient_accumulation_steps="1" \
|
||||
--logging_steps="25" \
|
||||
--learning_rate="1e-6" \
|
||||
--warmup_steps="500" \
|
||||
--evaluation_strategy="steps" \
|
||||
--eval_steps="1000" \
|
||||
--save_strategy="steps" \
|
||||
--save_steps="1000" \
|
||||
--generation_max_length="225" \
|
||||
--length_column_name="input_length" \
|
||||
--max_duration_in_seconds="30" \
|
||||
--text_column_name="sentence" \
|
||||
--freeze_feature_encoder="False" \
|
||||
--report_to="tensorboard" \
|
||||
--metric_for_best_model="wer" \
|
||||
--greater_is_better="False" \
|
||||
--load_best_model_at_end \
|
||||
--gradient_checkpointing \
|
||||
--fp16 \
|
||||
--overwrite_output_dir \
|
||||
--do_train \
|
||||
--do_eval \
|
||||
--predict_with_generate \
|
||||
--do_normalize_eval \
|
||||
--streaming="False" \
|
||||
--use_auth_token \
|
||||
--push_to_hub
|
||||
40
run.sh
Normal file
40
run.sh
Normal file
@@ -0,0 +1,40 @@
|
||||
deepspeed run_speech_recognition_seq2seq.py \
|
||||
--deepspeed="ds_config.json" \
|
||||
--model_name_or_path="openai/whisper-large-v3" \
|
||||
--dataset_name="mozilla-foundation/common_voice_13_0" \
|
||||
--dataset_config_name="cs" \
|
||||
--language="czech" \
|
||||
--train_split_name="train+validation" \
|
||||
--eval_split_name="test" \
|
||||
--max_steps="5000" \
|
||||
--output_dir="./" \
|
||||
--per_device_train_batch_size="62" \
|
||||
--per_device_eval_batch_size="16" \
|
||||
--gradient_accumulation_steps="1" \
|
||||
--logging_steps="25" \
|
||||
--learning_rate="1e-5" \
|
||||
--warmup_steps="500" \
|
||||
--evaluation_strategy="steps" \
|
||||
--eval_steps="1000" \
|
||||
--save_strategy="steps" \
|
||||
--save_steps="1000" \
|
||||
--do_lower_case="False" \
|
||||
--generation_max_length="225" \
|
||||
--preprocessing_num_workers="16" \
|
||||
--length_column_name="input_length" \
|
||||
--max_duration_in_seconds="30" \
|
||||
--text_column_name="sentence" \
|
||||
--freeze_feature_encoder="False" \
|
||||
--report_to="tensorboard" \
|
||||
--metric_for_best_model="wer" \
|
||||
--greater_is_better="False" \
|
||||
--load_best_model_at_end \
|
||||
--gradient_checkpointing \
|
||||
--group_by_length \
|
||||
--fp16 \
|
||||
--overwrite_output_dir \
|
||||
--do_train \
|
||||
--do_eval \
|
||||
--predict_with_generate \
|
||||
--use_auth_token \
|
||||
--push_to_hub
|
||||
625
run_speech_recognition_seq2seq.py
Normal file
625
run_speech_recognition_seq2seq.py
Normal file
@@ -0,0 +1,625 @@
|
||||
#!/usr/bin/env python
|
||||
# coding=utf-8
|
||||
# Copyright 2021 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.
|
||||
"""
|
||||
# You can also adapt this script on your own sequence to sequence speech
|
||||
# recognition task. Pointers for this are left as comments.
|
||||
|
||||
import logging
|
||||
import os
|
||||
import sys
|
||||
import warnings
|
||||
from dataclasses import dataclass, field
|
||||
from typing import Any, Dict, List, Optional, Union
|
||||
|
||||
import datasets
|
||||
import evaluate
|
||||
import torch
|
||||
from datasets import DatasetDict, load_dataset
|
||||
|
||||
import transformers
|
||||
from transformers import (
|
||||
AutoConfig,
|
||||
AutoFeatureExtractor,
|
||||
AutoModelForSpeechSeq2Seq,
|
||||
AutoProcessor,
|
||||
AutoTokenizer,
|
||||
HfArgumentParser,
|
||||
Seq2SeqTrainer,
|
||||
Seq2SeqTrainingArguments,
|
||||
set_seed,
|
||||
)
|
||||
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("4.36.0.dev0")
|
||||
|
||||
require_version("datasets>=1.18.0", "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)."},
|
||||
)
|
||||
token: str = field(
|
||||
default=None,
|
||||
metadata={
|
||||
"help": (
|
||||
"The token to use as HTTP bearer authorization for remote files. If not specified, will use the token "
|
||||
"generated when running `huggingface-cli login` (stored in `~/.huggingface`)."
|
||||
)
|
||||
},
|
||||
)
|
||||
use_auth_token: bool = field(
|
||||
default=None,
|
||||
metadata={
|
||||
"help": "The `use_auth_token` argument is deprecated and will be removed in v4.34. Please use `token` instead."
|
||||
},
|
||||
)
|
||||
trust_remote_code: bool = field(
|
||||
default=False,
|
||||
metadata={
|
||||
"help": (
|
||||
"Whether or not to allow for custom models defined on the Hub in their own modeling files. This option"
|
||||
"should only be set to `True` for repositories you trust and in which you have read the code, as it will "
|
||||
"execute code present on the Hub on your local machine."
|
||||
)
|
||||
},
|
||||
)
|
||||
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."}
|
||||
)
|
||||
apply_spec_augment: bool = field(
|
||||
default=False,
|
||||
metadata={
|
||||
"help": "Whether to apply *SpecAugment* data augmentation to the input features. This is currently only relevant for Wav2Vec2, HuBERT, WavLM and Whisper models."
|
||||
},
|
||||
)
|
||||
|
||||
|
||||
@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)."}
|
||||
)
|
||||
overwrite_cache: bool = field(
|
||||
default=False, metadata={"help": "Overwrite the cached training and evaluation sets"}
|
||||
)
|
||||
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 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"}
|
||||
)
|
||||
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"
|
||||
)
|
||||
},
|
||||
)
|
||||
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=True,
|
||||
metadata={"help": "Whether the target text should be lower cased."},
|
||||
)
|
||||
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."},
|
||||
)
|
||||
|
||||
|
||||
@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.
|
||||
forward_attention_mask (`bool`)
|
||||
Whether to return attention_mask.
|
||||
"""
|
||||
|
||||
processor: Any
|
||||
decoder_start_token_id: int
|
||||
forward_attention_mask: bool
|
||||
|
||||
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")
|
||||
|
||||
if self.forward_attention_mask:
|
||||
batch["attention_mask"] = torch.LongTensor([feature["attention_mask"] for feature in features])
|
||||
|
||||
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 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()
|
||||
|
||||
if model_args.use_auth_token is not None:
|
||||
warnings.warn(
|
||||
"The `use_auth_token` argument is deprecated and will be removed in v4.34. Please use `token` instead.",
|
||||
FutureWarning,
|
||||
)
|
||||
if model_args.token is not None:
|
||||
raise ValueError("`token` and `use_auth_token` are both specified. Please set only the argument `token`.")
|
||||
model_args.token = model_args.use_auth_token
|
||||
|
||||
# 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", 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: {training_args.parallel_mode.value == 'distributed'}, 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 = DatasetDict()
|
||||
|
||||
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,
|
||||
cache_dir=model_args.cache_dir,
|
||||
token=model_args.token,
|
||||
)
|
||||
|
||||
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,
|
||||
cache_dir=model_args.cache_dir,
|
||||
token=model_args.token,
|
||||
)
|
||||
|
||||
if data_args.audio_column_name not in next(iter(raw_datasets.values())).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(next(iter(raw_datasets.values())).column_names)}."
|
||||
)
|
||||
|
||||
if data_args.text_column_name not in next(iter(raw_datasets.values())).column_names:
|
||||
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(next(iter(raw_datasets.values())).column_names)}."
|
||||
)
|
||||
|
||||
# 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,
|
||||
token=model_args.token,
|
||||
trust_remote_code=model_args.trust_remote_code,
|
||||
)
|
||||
|
||||
config.update({"forced_decoder_ids": model_args.forced_decoder_ids, "suppress_tokens": model_args.suppress_tokens})
|
||||
|
||||
# SpecAugment for whisper models
|
||||
if getattr(config, "model_type", None) == "whisper":
|
||||
config.update({"apply_spec_augment": model_args.apply_spec_augment})
|
||||
|
||||
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,
|
||||
token=model_args.token,
|
||||
trust_remote_code=model_args.trust_remote_code,
|
||||
)
|
||||
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,
|
||||
token=model_args.token,
|
||||
trust_remote_code=model_args.trust_remote_code,
|
||||
)
|
||||
model = AutoModelForSpeechSeq2Seq.from_pretrained(
|
||||
model_args.model_name_or_path,
|
||||
config=config,
|
||||
cache_dir=model_args.cache_dir,
|
||||
revision=model_args.model_revision,
|
||||
token=model_args.token,
|
||||
trust_remote_code=model_args.trust_remote_code,
|
||||
)
|
||||
|
||||
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()
|
||||
model.model.encoder.gradient_checkpointing = False
|
||||
|
||||
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
|
||||
num_workers = data_args.preprocessing_num_workers
|
||||
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
|
||||
# if SpecAugment is used for whisper models, return attention_mask to guide the mask along time axis
|
||||
forward_attention_mask = (
|
||||
getattr(config, "model_type", None) == "whisper"
|
||||
and getattr(config, "apply_spec_augment", False)
|
||||
and getattr(config, "mask_time_prob", 0) > 0
|
||||
)
|
||||
|
||||
if data_args.max_train_samples is not None:
|
||||
raw_datasets["train"] = 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"].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"], return_attention_mask=forward_attention_mask
|
||||
)
|
||||
# process audio length
|
||||
batch[model_input_name] = inputs.get(model_input_name)[0]
|
||||
batch["input_length"] = len(sample["array"])
|
||||
if forward_attention_mask:
|
||||
batch["attention_mask"] = inputs.get("attention_mask")[0]
|
||||
|
||||
# process targets
|
||||
input_str = batch[text_column_name].lower() if do_lower_case else batch[text_column_name]
|
||||
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=next(iter(raw_datasets.values())).column_names,
|
||||
num_proc=data_args.preprocessing_num_workers,
|
||||
desc="preprocess train dataset",
|
||||
)
|
||||
|
||||
# filter data that is shorter than min_input_length or longer than
|
||||
# max_input_length
|
||||
def is_audio_in_length_range(length):
|
||||
return length > min_input_length and length < max_input_length
|
||||
|
||||
vectorized_datasets = vectorized_datasets.filter(
|
||||
is_audio_in_length_range,
|
||||
num_proc=num_workers,
|
||||
input_columns=["input_length"],
|
||||
)
|
||||
|
||||
# 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:
|
||||
cache = {k: v.cache_files for k, v in vectorized_datasets.items()}
|
||||
logger.info(f"Data preprocessing finished. Files cached at {cache}.")
|
||||
return
|
||||
|
||||
# 8. Load Metric
|
||||
metric = evaluate.load("wer")
|
||||
|
||||
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)
|
||||
|
||||
wer = metric.compute(predictions=pred_str, references=label_str)
|
||||
|
||||
return {"wer": wer}
|
||||
|
||||
# 9. Create a single speech processor
|
||||
# make sure all processes wait until data is saved
|
||||
with training_args.main_process_first():
|
||||
# only the main process saves them
|
||||
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,
|
||||
forward_attention_mask=forward_attention_mask,
|
||||
)
|
||||
|
||||
# 11. 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,
|
||||
)
|
||||
|
||||
# 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
|
||||
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()
|
||||
|
||||
# 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,
|
||||
)
|
||||
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)
|
||||
|
||||
# 14. Write Training Stats
|
||||
kwargs = {"finetuned_from": model_args.model_name_or_path, "tasks": "automatic-speech-recognition"}
|
||||
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_args"] = data_args.dataset_config_name
|
||||
kwargs["dataset"] = f"{data_args.dataset_name} {data_args.dataset_config_name}"
|
||||
else:
|
||||
kwargs["dataset"] = data_args.dataset_name
|
||||
|
||||
if training_args.push_to_hub:
|
||||
trainer.push_to_hub(**kwargs)
|
||||
else:
|
||||
trainer.create_model_card(**kwargs)
|
||||
|
||||
return results
|
||||
|
||||
|
||||
if __name__ == "__main__":
|
||||
main()
|
||||
629
run_speech_recognition_seq2seq_streaming.py
Normal file
629
run_speech_recognition_seq2seq_streaming.py
Normal file
@@ -0,0 +1,629 @@
|
||||
#!/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
|
||||
|
||||
if training_args.do_train:
|
||||
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]
|
||||
# filtering step to only evaluate the samples that correspond to non-zero references:
|
||||
pred_str = [pred_str[i] for i in range(len(pred_str)) if len(label_str[i]) > 0]
|
||||
label_str = [label_str[i] for i in range(len(label_str)) if len(label_str[i]) > 0]
|
||||
|
||||
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.split('-')[0]
|
||||
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()
|
||||
@@ -0,0 +1,3 @@
|
||||
version https://git-lfs.github.com/spec/v1
|
||||
oid sha256:2e2ac8a8a9e1e90d6627f516a852d953dd162e6a2676fddf5939abb3da2ef7e6
|
||||
size 5161
|
||||
@@ -0,0 +1,3 @@
|
||||
version https://git-lfs.github.com/spec/v1
|
||||
oid sha256:e16cb63fb9f0e03fa4fcd079db81ecda37373375b27dcae475d73cbeb1a1ea8f
|
||||
size 5007
|
||||
@@ -0,0 +1,3 @@
|
||||
version https://git-lfs.github.com/spec/v1
|
||||
oid sha256:9102a6f3df4e711d987b5d35c62d3b85384b52d2f706df8d3976edb09f1804dc
|
||||
size 38336
|
||||
139
special_tokens_map.json
Normal file
139
special_tokens_map.json
Normal file
@@ -0,0 +1,139 @@
|
||||
{
|
||||
"additional_special_tokens": [
|
||||
"<|startoftranscript|>",
|
||||
"<|en|>",
|
||||
"<|zh|>",
|
||||
"<|de|>",
|
||||
"<|es|>",
|
||||
"<|ru|>",
|
||||
"<|ko|>",
|
||||
"<|fr|>",
|
||||
"<|ja|>",
|
||||
"<|pt|>",
|
||||
"<|tr|>",
|
||||
"<|pl|>",
|
||||
"<|ca|>",
|
||||
"<|nl|>",
|
||||
"<|ar|>",
|
||||
"<|sv|>",
|
||||
"<|it|>",
|
||||
"<|id|>",
|
||||
"<|hi|>",
|
||||
"<|fi|>",
|
||||
"<|vi|>",
|
||||
"<|he|>",
|
||||
"<|uk|>",
|
||||
"<|el|>",
|
||||
"<|ms|>",
|
||||
"<|cs|>",
|
||||
"<|ro|>",
|
||||
"<|da|>",
|
||||
"<|hu|>",
|
||||
"<|ta|>",
|
||||
"<|no|>",
|
||||
"<|th|>",
|
||||
"<|ur|>",
|
||||
"<|hr|>",
|
||||
"<|bg|>",
|
||||
"<|lt|>",
|
||||
"<|la|>",
|
||||
"<|mi|>",
|
||||
"<|ml|>",
|
||||
"<|cy|>",
|
||||
"<|sk|>",
|
||||
"<|te|>",
|
||||
"<|fa|>",
|
||||
"<|lv|>",
|
||||
"<|bn|>",
|
||||
"<|sr|>",
|
||||
"<|az|>",
|
||||
"<|sl|>",
|
||||
"<|kn|>",
|
||||
"<|et|>",
|
||||
"<|mk|>",
|
||||
"<|br|>",
|
||||
"<|eu|>",
|
||||
"<|is|>",
|
||||
"<|hy|>",
|
||||
"<|ne|>",
|
||||
"<|mn|>",
|
||||
"<|bs|>",
|
||||
"<|kk|>",
|
||||
"<|sq|>",
|
||||
"<|sw|>",
|
||||
"<|gl|>",
|
||||
"<|mr|>",
|
||||
"<|pa|>",
|
||||
"<|si|>",
|
||||
"<|km|>",
|
||||
"<|sn|>",
|
||||
"<|yo|>",
|
||||
"<|so|>",
|
||||
"<|af|>",
|
||||
"<|oc|>",
|
||||
"<|ka|>",
|
||||
"<|be|>",
|
||||
"<|tg|>",
|
||||
"<|sd|>",
|
||||
"<|gu|>",
|
||||
"<|am|>",
|
||||
"<|yi|>",
|
||||
"<|lo|>",
|
||||
"<|uz|>",
|
||||
"<|fo|>",
|
||||
"<|ht|>",
|
||||
"<|ps|>",
|
||||
"<|tk|>",
|
||||
"<|nn|>",
|
||||
"<|mt|>",
|
||||
"<|sa|>",
|
||||
"<|lb|>",
|
||||
"<|my|>",
|
||||
"<|bo|>",
|
||||
"<|tl|>",
|
||||
"<|mg|>",
|
||||
"<|as|>",
|
||||
"<|tt|>",
|
||||
"<|haw|>",
|
||||
"<|ln|>",
|
||||
"<|ha|>",
|
||||
"<|ba|>",
|
||||
"<|jw|>",
|
||||
"<|su|>",
|
||||
"<|yue|>",
|
||||
"<|translate|>",
|
||||
"<|transcribe|>",
|
||||
"<|startoflm|>",
|
||||
"<|startofprev|>",
|
||||
"<|nospeech|>",
|
||||
"<|notimestamps|>"
|
||||
],
|
||||
"bos_token": {
|
||||
"content": "<|endoftext|>",
|
||||
"lstrip": false,
|
||||
"normalized": false,
|
||||
"rstrip": false,
|
||||
"single_word": false
|
||||
},
|
||||
"eos_token": {
|
||||
"content": "<|endoftext|>",
|
||||
"lstrip": false,
|
||||
"normalized": false,
|
||||
"rstrip": false,
|
||||
"single_word": false
|
||||
},
|
||||
"pad_token": {
|
||||
"content": "<|endoftext|>",
|
||||
"lstrip": false,
|
||||
"normalized": false,
|
||||
"rstrip": false,
|
||||
"single_word": false
|
||||
},
|
||||
"unk_token": {
|
||||
"content": "<|endoftext|>",
|
||||
"lstrip": false,
|
||||
"normalized": false,
|
||||
"rstrip": false,
|
||||
"single_word": false
|
||||
}
|
||||
}
|
||||
114903
tokenizer.json
Normal file
114903
tokenizer.json
Normal file
File diff suppressed because it is too large
Load Diff
12996
tokenizer_config.json
Normal file
12996
tokenizer_config.json
Normal file
File diff suppressed because it is too large
Load Diff
3
training_args.bin
Normal file
3
training_args.bin
Normal file
@@ -0,0 +1,3 @@
|
||||
version https://git-lfs.github.com/spec/v1
|
||||
oid sha256:1adac70f1f1f82a14f7164910af93cebe7c58a829359d4b895c9772e839d3f49
|
||||
size 5883
|
||||
1
vocab.json
Normal file
1
vocab.json
Normal file
File diff suppressed because one or more lines are too long
Reference in New Issue
Block a user