ales commited on
Commit
8d89806
1 Parent(s): fc6cd04
src/readme.md ADDED
@@ -0,0 +1,129 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ ## Description
2
+
3
+ Fine-tuning [OpenAI Whisper](https://github.com/openai/whisper) model for Belarusian language during
4
+ [Whisper fine-tuning Event](https://github.com/huggingface/community-events/tree/main/whisper-fine-tuning-event)
5
+ hosted by HuggingFace x Lambda.
6
+
7
+ The code in this repository is a modified version of code from
8
+ [Whisper fine-tuning Event](https://github.com/huggingface/community-events/tree/main/whisper-fine-tuning-event) repo.
9
+
10
+ ## Tips:
11
+ * start with a port worwarding to monitor Tensorboard logs on local computer:
12
+ ```
13
+ ssh <remote-address> -L <local_port>:localhost:<remote_tensorboard_port>
14
+ ```
15
+ * Train with redirecting output to a file using `tee`:
16
+ ```
17
+ source src/run.sh 2>&1 | tee train_run_<run_number>.log
18
+ ```
19
+
20
+ ## Fine-tuning todos:
21
+ * perform evaluation of fine-tuned model on CommonVoice test set
22
+ * Learning rate:
23
+ * max learning rate is not the same as LR passed as a parameter to training script. it is actually lower.
24
+ * when resuming training, LR scheduling behaves incorrectly
25
+ * check exact sizes of train, eval, test sets of CommonVoice 11
26
+
27
+ ## Resuming training from exising checkpoint
28
+ When resuming training from existing checkpoint:
29
+ * it's better to save all `checkpoint-\d+` dirs. better not to rely on data saved to `output_dir` because:
30
+ * not all data is saved to `output_dir`. e.g. following files are not saved to `output_dir`:
31
+ `optimizer.pt`, `rng_state.pth`, `scaler.pt`, `scheduler.pt`. so can't resume training in a correct way from
32
+ data saved to `output_dir`
33
+ * when resuming training from `output_dir` as a checkpoint dir, model saved to `output_dir` can be worse than
34
+ previously save (need to investifate further. but such happened already)
35
+ * learning rate gets reset if passing same parameter value to training script as in previour run.<br>
36
+ need to provide learning rate from the last step of previous run to continue
37
+ training in a correct way
38
+ * however even if passing learning rate from the last step, in the new run it has different value than expected
39
+ * probably because last checkpont was chosen incorrectly
40
+ * or learning rate is treated as a starting learning rate at step 0 and not on step X (where we resume).<br>
41
+ need to try to pass same LR that was passes as a starting LR to the very first run
42
+ * it's unclear whether decision on saving current model
43
+ is made by comparing current metrics with metrics of the best checkpoint. I guess model with worse performance
44
+ will not overwrite best model checkpoint already exising in the output dir, but need to double check.
45
+ * we can set `ignore_data_skip=True` Training argument not to
46
+ skip data items already passed to a model - that will save time on data loads.
47
+ * it's unclear whether order of input items in the train set (that is shuffled) will be the same
48
+ across multiple reruns - i.e. it's unclear whether sampling is the same across reruns.
49
+ * if the sampling is the same across reruns, `ignore_data_skip=True` will lead to same items been passed to a model
50
+ in current run. it's OK if previous run ended with large step value on the last epoch.
51
+ if not, the same elements from the same epoch will be passed to a model again.
52
+
53
+ ## Questions:
54
+ * What checkpoint (best, I guess) is saved in the `output_dir`?
55
+ How is it overwritten when resuming training from existing checkpoint?
56
+ * does `ShuffleCallback` work with StreamingDataset? it reshuffles data `on_epoch_begin()`,
57
+ but does StreamingDataset have any epochs?
58
+
59
+ ### Prepended tokens
60
+ * Why are there following lines in Data Collator?
61
+ ```python
62
+ # if bos token is appended in previous tokenization step,
63
+ # cut bos token here as it's append later anyways
64
+ if (labels[:, 0] == self.decoder_start_token_id).all().cpu().item():
65
+ labels = labels[:, 1:]
66
+ ```
67
+ * `tokenizer.bos_token_id` vs `model.config.decoder_start_token_id`.<br>
68
+ which one to pass to Data Collator as `decoder_start_token_id` parameter?
69
+ * Answer:
70
+ * In this case, the two are equivalent. You can verify this:
71
+ ```python
72
+ print(tokenizer.bos_token_id)
73
+ print(model.config.decoder_start_token_id)
74
+ ```
75
+
76
+ * Print Output:
77
+ ```
78
+ <|startoftranscript|>
79
+ <|startoftranscript|>
80
+ ```
81
+
82
+ * Technically speaking, the decoder_start_token_id is the correct convention here. Before starting generating any tokens, we initialise the generate method with a starting token, which is the decoder_start_token_id.
83
+ See: https://huggingface.co/blog/how-to-generate. The decoder_start_token_id corresponds to the initial context word sequence, and is the zero'th token generated.
84
+
85
+ * We remove this token from the encoded labels in the data collator because we always set the zero'th generated token to the decoder_start_token_id. If we leave the decoder_start_token_id as part of the label sequence, then we'll predict the decoder_start_token_id as the zero'th token, and again as the first token! Because we're always forcing it as the zero'th token, we don't need to predict it as the first token, and so we remove it from the target lables
86
+
87
+ * These tokens are not forced in the generation process, and so we don't cut them in the data collator. We need to provide them to the model as target labels so that the model can learn the correct tasks from our data
88
+
89
+ * The tokens correspond to the audio language, task (translate or transcribe) and whether to predict timestamps
90
+
91
+ * We need to tell the model what language the audio corresponds to and what task it's performing during fine-tuning. This way, it learns what audio corresponds to what language, and the difference between transcribing audio vs translating it
92
+
93
+ ## Notes:
94
+ * using CommonVoice 11 dataset in a streaming way.<br>
95
+ use `streaming=True` for train & validation & test.<br>
96
+ as an alternative, we can use `streaming=False` for validation & test sets to save time on data processing.
97
+ but the size of validation and test sets are unknown (need to check).
98
+ it's likely they are going to be large - thus pre-download of these sets might not reduce
99
+ overall fine-tuning time compared to streaming mode.
100
+ * size of train set is ~370'000 audiofiles. if using `batch_size=64`, then
101
+ 1 epoch will have ~5782 steps. <br>
102
+ Because of `--eval_steps="1000"` will use `--max_steps="6000"` instead of `--max_steps="5800"`
103
+ to have evaluation metrics computed in the end of training.
104
+ * if using Google Colab, need to execute `sudo chmod -R 777 .git` inside hf repo to
105
+ to set right permissions to be able to push trained models to HuggingFace Hub
106
+ * Whispers BasicTextNormalizer splits words containing apostrophe:
107
+ ```python
108
+ > from transformers.models.whisper.english_normalizer import BasicTextNormalizer
109
+ > normalizer = BasicTextNormalizer()
110
+ > normalizer("раз'яднаць")
111
+ 'раз яднаць'
112
+ ```
113
+ * That's why `BelarusianTextNormalizer` (edited version of `BasicTextNormalizer`) was added to training script:
114
+ ```python
115
+ > from run_speech_recognition_seq2seq_streaming import BelarusianTextNormalizer
116
+ > normalizer_be = BelarusianTextNormalizer()
117
+ > normalizer_be("раз'яднаць")
118
+ "раз'яднаць"
119
+ ```
120
+ * Need to set `use_cache` to False since we're using gradient checkpointing, and the two are incompatible
121
+ * Default Linear scheduler is used
122
+ * Default Adam optimizer is used
123
+ * To save memory (and increase either model or batch_size) can experiment with:
124
+ * using Adafactor instead of Adam.
125
+ Adam requires two optimiser params per one model param, but Adafactor uses only one.
126
+ > A word of caution: Adafactor is untested for fine-tuning Whisper,
127
+ so we are unsure sure how Adafactor performance compares to Adam!
128
+ * using Adam 8bit from `bitsandbytes` module.
129
+ need to provide `optim="adamw_bnb_8bit"` param to `Seq2SeqTrainingArguments`
src/requirements.txt ADDED
@@ -0,0 +1,9 @@
 
 
 
 
 
 
 
 
 
 
1
+ torch>=1.7
2
+ torchaudio
3
+ git+https://github.com/huggingface/transformers
4
+ git+https://github.com/huggingface/datasets
5
+ librosa
6
+ jiwer
7
+ evaluate>=0.3.0
8
+ more-itertools
9
+ tensorboard
src/run.sh ADDED
@@ -0,0 +1,43 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ python src/run_speech_recognition_seq2seq_streaming.py \
2
+ --model_name_or_path="openai/whisper-small" \
3
+ --dataset_name="mozilla-foundation/common_voice_11_0" \
4
+ --dataset_config_name="be" \
5
+ --language="be" \
6
+ --train_split_name="train" \
7
+ --eval_split_name="validation" \
8
+ --model_index_name="Whisper Small Belarusian" \
9
+ \
10
+ --max_steps="12000" \
11
+ --output_dir="./" \
12
+ --per_device_train_batch_size="64" \
13
+ --per_device_eval_batch_size="64" \
14
+ --logging_steps="50" \
15
+ --logging_first_step \
16
+ --learning_rate="1e-4" \
17
+ --warmup_steps="500" \
18
+ --evaluation_strategy="steps" \
19
+ --eval_steps="1000" \
20
+ --save_strategy="steps" \
21
+ --save_steps="1000" \
22
+ --gradient_checkpointing \
23
+ --fp16 \
24
+ \
25
+ --shuffle_buffer_size="500" \
26
+ --generation_max_length="225" \
27
+ --max_duration_in_seconds="30" \
28
+ --text_column_name="sentence" \
29
+ --freeze_feature_encoder="False" \
30
+ --report_to="tensorboard" \
31
+ --metric_for_best_model="wer" \
32
+ --greater_is_better="False" \
33
+ --load_best_model_at_end \
34
+ \
35
+ --do_train \
36
+ --do_eval \
37
+ --ignore_data_skip \
38
+ --predict_with_generate \
39
+ --do_normalize_eval \
40
+ --streaming \
41
+ --use_auth_token \
42
+ --push_to_hub \
43
+ --hub_model_id="ales/whisper-small-belarusian"
src/run_debug.sh ADDED
@@ -0,0 +1,44 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ python src/run_speech_recognition_seq2seq_streaming.py \
2
+ --model_name_or_path="openai/whisper-tiny" \
3
+ --dataset_name="mozilla-foundation/common_voice_11_0" \
4
+ --dataset_config_name="be" \
5
+ --language="be" \
6
+ --train_split_name="train" \
7
+ --eval_split_name="validation" \
8
+ --model_index_name="Whisper Tiny Belarusian" \
9
+ \
10
+ --max_steps="500" \
11
+ --max_eval_samples="64" \
12
+ --output_dir="./" \
13
+ --per_device_train_batch_size="32" \
14
+ --per_device_eval_batch_size="32" \
15
+ --logging_steps="10" \
16
+ --logging_first_step \
17
+ --learning_rate="1e-4" \
18
+ --warmup_steps="10" \
19
+ --evaluation_strategy="steps" \
20
+ --eval_steps="10" \
21
+ --save_strategy="steps" \
22
+ --save_steps="10" \
23
+ --gradient_checkpointing \
24
+ --fp16 \
25
+ \
26
+ --shuffle_buffer_size="20" \
27
+ --generation_max_length="225" \
28
+ --max_duration_in_seconds="30" \
29
+ --text_column_name="sentence" \
30
+ --freeze_feature_encoder="False" \
31
+ --report_to="tensorboard" \
32
+ --metric_for_best_model="wer" \
33
+ --greater_is_better="False" \
34
+ --load_best_model_at_end \
35
+ \
36
+ --do_train \
37
+ --do_eval \
38
+ --ignore_data_skip \
39
+ --predict_with_generate \
40
+ --do_normalize_eval \
41
+ --streaming \
42
+ --use_auth_token \
43
+ --push_to_hub \
44
+ --hub_model_id="ales/whisper-tiny-be-test"
src/run_speech_recognition_seq2seq_streaming.py ADDED
@@ -0,0 +1,730 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ #!/usr/bin/env python
2
+ # coding=utf-8
3
+ # Copyright 2022 The HuggingFace Team. All rights reserved.
4
+ #
5
+ # Licensed under the Apache License, Version 2.0 (the "License");
6
+ # you may not use this file except in compliance with the License.
7
+ # You may obtain a copy of the License at
8
+ #
9
+ # http://www.apache.org/licenses/LICENSE-2.0
10
+ #
11
+ # Unless required by applicable law or agreed to in writing, software
12
+ # distributed under the License is distributed on an "AS IS" BASIS,
13
+ # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14
+ # See the License for the specific language governing permissions and
15
+ # limitations under the License.
16
+ """
17
+ Fine-tuning the library models for sequence to sequence speech recognition
18
+ with 🤗 Datasets' streaming mode.
19
+ """
20
+ # You can also adapt this script for your own sequence to sequence speech
21
+ # recognition task. Pointers for this are left as comments.
22
+
23
+ import logging
24
+ import os
25
+ import sys
26
+ import datetime
27
+ import re
28
+ import regex
29
+ import unicodedata
30
+ from dataclasses import dataclass, field
31
+ from typing import Any, Dict, List, Optional, Union, Iterable
32
+
33
+ import datasets
34
+ import torch
35
+ from datasets import DatasetDict, IterableDatasetDict, interleave_datasets, load_dataset
36
+ from torch.utils.data import IterableDataset
37
+
38
+ import evaluate
39
+ import transformers
40
+ from transformers import (
41
+ AutoConfig,
42
+ AutoFeatureExtractor,
43
+ AutoModelForSpeechSeq2Seq,
44
+ AutoProcessor,
45
+ AutoTokenizer,
46
+ HfArgumentParser,
47
+ Seq2SeqTrainer,
48
+ Seq2SeqTrainingArguments,
49
+ TrainerCallback,
50
+ set_seed,
51
+ )
52
+ from transformers.trainer_pt_utils import IterableDatasetShard
53
+ from transformers.trainer_utils import get_last_checkpoint, is_main_process
54
+ from transformers.utils import check_min_version, send_example_telemetry
55
+ from transformers.utils.versions import require_version
56
+
57
+
58
+ # Will error if the minimal version of Transformers is not installed. Remove at your own risks.
59
+ check_min_version("4.25.0.dev0")
60
+
61
+ require_version("datasets>=1.18.2", "To fix: pip install -r examples/pytorch/speech-recognition/requirements.txt")
62
+
63
+ logger = logging.getLogger(__name__)
64
+
65
+
66
+ @dataclass
67
+ class ModelArguments:
68
+ """
69
+ Arguments pertaining to which model/config/tokenizer we are going to fine-tune from.
70
+ """
71
+
72
+ model_name_or_path: str = field(
73
+ metadata={"help": "Path to pretrained model or model identifier from huggingface.co/models"}
74
+ )
75
+ config_name: Optional[str] = field(
76
+ default=None, metadata={"help": "Pretrained config name or path if not the same as model_name"}
77
+ )
78
+ tokenizer_name: Optional[str] = field(
79
+ default=None, metadata={"help": "Pretrained tokenizer name or path if not the same as model_name"}
80
+ )
81
+ feature_extractor_name: Optional[str] = field(
82
+ default=None, metadata={"help": "feature extractor name or path if not the same as model_name"}
83
+ )
84
+ cache_dir: Optional[str] = field(
85
+ default=None,
86
+ metadata={"help": "Where to store the pretrained models downloaded from huggingface.co"},
87
+ )
88
+ use_fast_tokenizer: bool = field(
89
+ default=True,
90
+ metadata={"help": "Whether to use one of the fast tokenizer (backed by the tokenizers library) or not."},
91
+ )
92
+ model_revision: str = field(
93
+ default="main",
94
+ metadata={"help": "The specific model version to use (can be a branch name, tag name or commit id)."},
95
+ )
96
+ use_auth_token: bool = field(
97
+ default=False,
98
+ metadata={
99
+ "help": (
100
+ "Will use the token generated when running `huggingface-cli login` (necessary to use this script "
101
+ "with private models)."
102
+ )
103
+ },
104
+ )
105
+ freeze_feature_encoder: bool = field(
106
+ default=True, metadata={"help": "Whether to freeze the feature encoder layers of the model."}
107
+ )
108
+ freeze_encoder: bool = field(
109
+ default=False, metadata={"help": "Whether to freeze the entire encoder of the seq2seq model."}
110
+ )
111
+ forced_decoder_ids: List[List[int]] = field(
112
+ default=None,
113
+ metadata={
114
+ "help": (
115
+ "A list of pairs of integers which indicates a mapping from generation indices to token indices "
116
+ "that will be forced before sampling. For example, [[0, 123]] means the first generated token "
117
+ "will always be a token of index 123."
118
+ )
119
+ },
120
+ )
121
+ suppress_tokens: List[int] = field(
122
+ default=None, metadata={"help": "A list of tokens that will be suppressed at generation."}
123
+ )
124
+ model_index_name: str = field(default=None, metadata={"help": "Pretty name for the model card."})
125
+
126
+
127
+ @dataclass
128
+ class DataTrainingArguments:
129
+ """
130
+ Arguments pertaining to what data we are going to input our model for training and eval.
131
+ """
132
+
133
+ dataset_name: str = field(
134
+ default=None, metadata={"help": "The name of the dataset to use (via the datasets library)."}
135
+ )
136
+ dataset_config_name: Optional[str] = field(
137
+ default=None, metadata={"help": "The configuration name of the dataset to use (via the datasets library)."}
138
+ )
139
+ max_train_samples: Optional[int] = field(
140
+ default=None,
141
+ metadata={
142
+ "help": (
143
+ "For debugging purposes or quicker training, truncate the number of training examples to this "
144
+ "value if set."
145
+ )
146
+ },
147
+ )
148
+ max_eval_samples: Optional[int] = field(
149
+ default=None,
150
+ metadata={
151
+ "help": (
152
+ "For debugging purposes or quicker training, truncate the number of evaluation examples to this "
153
+ "value if set."
154
+ )
155
+ },
156
+ )
157
+ audio_column_name: str = field(
158
+ default="audio",
159
+ metadata={"help": "The name of the dataset column containing the audio data. Defaults to 'audio'"},
160
+ )
161
+ text_column_name: str = field(
162
+ default="text",
163
+ metadata={"help": "The name of the dataset column containing the text data. Defaults to 'text'"},
164
+ )
165
+ max_duration_in_seconds: float = field(
166
+ default=20.0,
167
+ metadata={
168
+ "help": (
169
+ "Truncate audio files that are longer than `max_duration_in_seconds` seconds to"
170
+ " 'max_duration_in_seconds`"
171
+ )
172
+ },
173
+ )
174
+ min_duration_in_seconds: float = field(
175
+ default=0.0, metadata={"help": "Filter audio files that are shorter than `min_duration_in_seconds` seconds"}
176
+ )
177
+ train_split_name: str = field(
178
+ default="train",
179
+ metadata={
180
+ "help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'"
181
+ },
182
+ )
183
+ eval_split_name: str = field(
184
+ default="test",
185
+ metadata={
186
+ "help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'"
187
+ },
188
+ )
189
+ do_lower_case: bool = field(
190
+ default=False,
191
+ metadata={"help": "Whether the target text should be lower cased."},
192
+ )
193
+ do_remove_punctuation: bool = field(
194
+ default=False,
195
+ metadata={"help": "Whether the target text should be striped of punctuation."},
196
+ )
197
+ do_normalize_eval: bool = field(
198
+ default=True,
199
+ metadata={"help": "Whether to normalise the references and predictions in the eval WER calculation."},
200
+ )
201
+ language: str = field(
202
+ default=None,
203
+ metadata={
204
+ "help": (
205
+ "Language for multilingual fine-tuning. This argument should be set for multilingual fine-tuning "
206
+ "only. For English speech recognition, it should be set to `None`."
207
+ )
208
+ },
209
+ )
210
+ task: str = field(
211
+ default="transcribe",
212
+ metadata={"help": "Task, either `transcribe` for speech recognition or `translate` for speech translation."},
213
+ )
214
+ shuffle_buffer_size: Optional[int] = field(
215
+ default=500,
216
+ metadata={
217
+ "help": (
218
+ "The number of streamed examples to download before shuffling them. The large the buffer, "
219
+ "the closer it is to real offline shuffling."
220
+ )
221
+ },
222
+ )
223
+ streaming: bool = field(
224
+ default=True,
225
+ metadata={"help": "Whether to use streaming mode to load and pre-process the data."},
226
+ )
227
+
228
+
229
+ class BelarusianTextNormalizer:
230
+ """
231
+ Based on transformers.models.whisper.english_normalizer.BasicTextNormalizer
232
+ but with support not to remove certain characters.
233
+ e.g. apostrophe (') - a symbol from Belarusian alphabet - was removed using BasicTextNormalizer.
234
+ """
235
+
236
+ def __init__(self, split_letters: bool = False):
237
+ self.split_letters = split_letters
238
+ self.allowed_symbols = ("'",)
239
+
240
+ @staticmethod
241
+ def clean(s: str, allowed_symbols: Iterable[str] = None):
242
+ """
243
+ Replace any other markers, symbols, punctuations with a space, keeping diacritics
244
+ """
245
+ if allowed_symbols is None:
246
+ allowed_symbols = []
247
+ res = "".join(" " if unicodedata.category(c)[0] in "MSP" and c not in allowed_symbols else c
248
+ for c in unicodedata.normalize("NFKC", s))
249
+ return res
250
+
251
+ def __call__(self, s: str):
252
+ s = s.lower()
253
+ s = re.sub(r"[<\[][^>\]]*[>\]]", "", s) # remove words between brackets
254
+ s = re.sub(r"\(([^)]+?)\)", "", s) # remove words between parenthesis
255
+ s = self.clean(s, allowed_symbols=self.allowed_symbols).lower()
256
+
257
+ if self.split_letters:
258
+ s = " ".join(regex.findall(r"\X", s, regex.U))
259
+
260
+ s = re.sub(r"\s+", " ", s) # replace any successive whitespace characters with a space
261
+
262
+ return s
263
+
264
+
265
+ @dataclass
266
+ class DataCollatorSpeechSeq2SeqWithPadding:
267
+ """
268
+ Data collator that will dynamically pad the inputs received.
269
+ Args:
270
+ processor ([`WhisperProcessor`])
271
+ The processor used for processing the data.
272
+ decoder_start_token_id (`int`)
273
+ The begin-of-sentence of the decoder.
274
+ """
275
+
276
+ processor: Any
277
+ decoder_start_token_id: int
278
+
279
+ def __call__(self, features: List[Dict[str, Union[List[int], torch.Tensor]]]) -> Dict[str, torch.Tensor]:
280
+ # split inputs and labels since they have to be of different lengths and need
281
+ # different padding methods
282
+ model_input_name = self.processor.model_input_names[0]
283
+ input_features = [{model_input_name: feature[model_input_name]} for feature in features]
284
+ label_features = [{"input_ids": feature["labels"]} for feature in features]
285
+
286
+ batch = self.processor.feature_extractor.pad(input_features, return_tensors="pt")
287
+
288
+ labels_batch = self.processor.tokenizer.pad(label_features, return_tensors="pt")
289
+
290
+ # replace padding with -100 to ignore loss correctly
291
+ labels = labels_batch["input_ids"].masked_fill(labels_batch.attention_mask.ne(1), -100)
292
+
293
+ # if bos token is appended in previous tokenization step,
294
+ # cut bos token here as it's append later anyways
295
+ if (labels[:, 0] == self.decoder_start_token_id).all().cpu().item():
296
+ labels = labels[:, 1:]
297
+
298
+ batch["labels"] = labels
299
+
300
+ return batch
301
+
302
+
303
+ def load_maybe_streaming_dataset(dataset_name, dataset_config_name, split="train", streaming=True, **kwargs):
304
+ """
305
+ Utility function to load a dataset in streaming mode. For datasets with multiple splits,
306
+ each split is loaded individually and then splits combined by taking alternating examples from
307
+ each (interleaving).
308
+ """
309
+ if "+" in split:
310
+ # load multiple splits separated by the `+` symbol with streaming mode
311
+ dataset_splits = [
312
+ load_dataset(dataset_name, dataset_config_name, split=split_name, streaming=streaming, **kwargs)
313
+ for split_name in split.split("+")
314
+ ]
315
+ # interleave multiple splits to form one dataset
316
+ interleaved_dataset = interleave_datasets(dataset_splits)
317
+ return interleaved_dataset
318
+ else:
319
+ # load a single split *with* streaming mode
320
+ dataset = load_dataset(dataset_name, dataset_config_name, split=split, streaming=streaming, **kwargs)
321
+ return dataset
322
+
323
+
324
+ def main():
325
+ # 1. Parse input arguments
326
+ # See all possible arguments in src/transformers/training_args.py
327
+ # or by passing the --help flag to this script.
328
+ # We now keep distinct sets of args, for a cleaner separation of concerns.
329
+ parser = HfArgumentParser((ModelArguments, DataTrainingArguments, Seq2SeqTrainingArguments))
330
+
331
+ if len(sys.argv) == 2 and sys.argv[1].endswith(".json"):
332
+ # If we pass only one argument to the script and it's the path to a json file,
333
+ # let's parse it to get our arguments.
334
+ model_args, data_args, training_args = parser.parse_json_file(json_file=os.path.abspath(sys.argv[1]))
335
+ else:
336
+ model_args, data_args, training_args = parser.parse_args_into_dataclasses()
337
+
338
+
339
+ # 2. Setup logging
340
+ now_str = datetime.datetime.now().strftime('%Y%m%d-%H%M%S')
341
+ logging.basicConfig(
342
+ format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
343
+ datefmt="%m/%d/%Y %H:%M:%S",
344
+ handlers=[
345
+ logging.StreamHandler(sys.stdout),
346
+ logging.FileHandler(filename=f'train_{now_str}.log', mode='w')
347
+ ],
348
+ )
349
+ log_level = training_args.get_process_log_level()
350
+ logger.setLevel(log_level)
351
+ datasets.utils.logging.set_verbosity(log_level)
352
+ transformers.utils.logging.set_verbosity(log_level)
353
+ transformers.utils.logging.enable_default_handler()
354
+ transformers.utils.logging.enable_explicit_format()
355
+
356
+ logger.setLevel(logging.INFO if is_main_process(training_args.local_rank) else logging.WARN)
357
+
358
+ # Log on each process the small summary:
359
+ logger.warning(
360
+ f"Process rank: {training_args.local_rank}, device: {training_args.device}, n_gpu: {training_args.n_gpu}"
361
+ f"distributed training: {bool(training_args.local_rank != -1)}, 16-bits training: {training_args.fp16}"
362
+ )
363
+ logger.info(f"Training/evaluation parameters {training_args}")
364
+
365
+ # Set the verbosity to info of the Transformers logger (on main process only):
366
+ if is_main_process(training_args.local_rank):
367
+ transformers.utils.logging.set_verbosity_info()
368
+ logger.info("Training/evaluation parameters %s", training_args)
369
+
370
+ # 3. Detecting last checkpoint and eventually continue from last checkpoint
371
+ last_checkpoint = None
372
+ if os.path.isdir(training_args.output_dir) and training_args.do_train and not training_args.overwrite_output_dir:
373
+ logger.info(f'output_dir already exists. will try to load last checkpoint.')
374
+
375
+ last_checkpoint = get_last_checkpoint(training_args.output_dir)
376
+ if last_checkpoint is not None:
377
+ if training_args.resume_from_checkpoint is None:
378
+ logger.info(
379
+ f"Checkpoint detected, resuming training at {last_checkpoint}. To avoid this behavior, change "
380
+ "the `--output_dir` or add `--overwrite_output_dir` to train from scratch."
381
+ )
382
+ else:
383
+ logger.info(f'Last checkpoint found at: {last_checkpoint}. Will ignore it and resume training '
384
+ f'from passed resume_from_checkpoint param: {training_args.resume_from_checkpoint}')
385
+ assert os.path.isdir(training_args.resume_from_checkpoint)
386
+ else:
387
+ logger.info('last_checkpoint is None. will try to read from training_args.resume_from_checkpoint')
388
+
389
+ if training_args.resume_from_checkpoint is not None and os.path.isdir(training_args.resume_from_checkpoint):
390
+ logger.info(f'Will resume training from passed resume_from_checkpoint param: '
391
+ f'{training_args.resume_from_checkpoint}')
392
+ else:
393
+ logger.info('last_checkpoint is None. resume_from_checkpoint is either None or not existing dir. '
394
+ 'will try to read from the model saved in the root of output_dir.')
395
+
396
+ dir_content = os.listdir(training_args.output_dir)
397
+ if len(dir_content) == 0:
398
+ logger.info('output_dir is empty. will start training from scratch.')
399
+ else:
400
+ model_fn = 'pytorch_model.bin'
401
+ if model_fn in dir_content:
402
+ logger.info(f'found {model_fn} inside output_dir. '
403
+ f'will continue training treating output_dir as a last checkpoint.')
404
+ last_checkpoint = training_args.output_dir
405
+ else:
406
+ allowed_dirs = ['.git', '.gitattributes', 'src']
407
+ unexpected_content = set(dir_content).difference(allowed_dirs)
408
+ if len(unexpected_content) > 0:
409
+ raise ValueError(
410
+ f'Could not find last_checkpoint, resume_from_checkpoint is either None '
411
+ 'or not existing dir, output_dir is non-empty but does not contain a model.'
412
+ 'Use --overwrite_output_dir to overcome. '
413
+ f'unexpected_content: {unexpected_content}'
414
+ )
415
+ else:
416
+ logger.info(f'dir is not empty, but contains only: {dir_content}. '
417
+ 'it is OK - will start training')
418
+
419
+
420
+ # Set seed before initializing model.
421
+ set_seed(training_args.seed)
422
+
423
+ # 4. Load dataset
424
+ raw_datasets = IterableDatasetDict() if data_args.streaming else DatasetDict()
425
+
426
+ if training_args.do_train:
427
+ raw_datasets["train"] = load_maybe_streaming_dataset(
428
+ data_args.dataset_name,
429
+ data_args.dataset_config_name,
430
+ split=data_args.train_split_name,
431
+ use_auth_token=True if model_args.use_auth_token else None,
432
+ streaming=data_args.streaming,
433
+ )
434
+
435
+ if training_args.do_eval:
436
+ raw_datasets["eval"] = load_maybe_streaming_dataset(
437
+ data_args.dataset_name,
438
+ data_args.dataset_config_name,
439
+ split=data_args.eval_split_name,
440
+ use_auth_token=True if model_args.use_auth_token else None,
441
+ streaming=data_args.streaming,
442
+ )
443
+
444
+ raw_datasets_features = list(next(iter(raw_datasets.values())).features.keys())
445
+
446
+ if data_args.audio_column_name not in raw_datasets_features:
447
+ raise ValueError(
448
+ f"--audio_column_name '{data_args.audio_column_name}' not found in dataset '{data_args.dataset_name}'. "
449
+ "Make sure to set `--audio_column_name` to the correct audio column - one of "
450
+ f"{', '.join(raw_datasets_features)}."
451
+ )
452
+
453
+ if data_args.text_column_name not in raw_datasets_features:
454
+ raise ValueError(
455
+ f"--text_column_name {data_args.text_column_name} not found in dataset '{data_args.dataset_name}'. "
456
+ "Make sure to set `--text_column_name` to the correct text column - one of "
457
+ f"{', '.join(raw_datasets_features)}."
458
+ )
459
+
460
+ # 5. Load pretrained model, tokenizer, and feature extractor
461
+ #
462
+ # Distributed training:
463
+ # The .from_pretrained methods guarantee that only one local process can concurrently
464
+ config = AutoConfig.from_pretrained(
465
+ model_args.config_name if model_args.config_name else model_args.model_name_or_path,
466
+ cache_dir=model_args.cache_dir,
467
+ revision=model_args.model_revision,
468
+ use_auth_token=True if model_args.use_auth_token else None,
469
+ )
470
+
471
+ config.update({"forced_decoder_ids": model_args.forced_decoder_ids, "suppress_tokens": model_args.suppress_tokens})
472
+
473
+ if training_args.gradient_checkpointing:
474
+ config.update({"use_cache": False})
475
+
476
+ feature_extractor = AutoFeatureExtractor.from_pretrained(
477
+ model_args.feature_extractor_name if model_args.feature_extractor_name else model_args.model_name_or_path,
478
+ cache_dir=model_args.cache_dir,
479
+ revision=model_args.model_revision,
480
+ use_auth_token=True if model_args.use_auth_token else None,
481
+ )
482
+ tokenizer = AutoTokenizer.from_pretrained(
483
+ model_args.tokenizer_name if model_args.tokenizer_name else model_args.model_name_or_path,
484
+ cache_dir=model_args.cache_dir,
485
+ use_fast=model_args.use_fast_tokenizer,
486
+ revision=model_args.model_revision,
487
+ use_auth_token=True if model_args.use_auth_token else None,
488
+ )
489
+ model = AutoModelForSpeechSeq2Seq.from_pretrained(
490
+ model_args.model_name_or_path,
491
+ config=config,
492
+ cache_dir=model_args.cache_dir,
493
+ revision=model_args.model_revision,
494
+ use_auth_token=True if model_args.use_auth_token else None,
495
+ )
496
+
497
+ if model.config.decoder_start_token_id is None:
498
+ raise ValueError("Make sure that `config.decoder_start_token_id` is correctly defined")
499
+
500
+ if model_args.freeze_feature_encoder:
501
+ model.freeze_feature_encoder()
502
+
503
+ if model_args.freeze_encoder:
504
+ model.freeze_encoder()
505
+
506
+ if data_args.language is not None:
507
+ # We only need to set the task id when the language is specified (i.e. in a multilingual setting)
508
+ tokenizer.set_prefix_tokens(language=data_args.language, task=data_args.task)
509
+
510
+ # 6. Explicitly resample speech dataset
511
+ raw_datasets = raw_datasets.cast_column(
512
+ data_args.audio_column_name, datasets.features.Audio(
513
+ sampling_rate=feature_extractor.sampling_rate,
514
+ mono=True
515
+ )
516
+ )
517
+
518
+ # 7. Preprocessing the datasets.
519
+ # We need to read the audio files as arrays and tokenize the targets.
520
+ max_input_length = data_args.max_duration_in_seconds * feature_extractor.sampling_rate
521
+ min_input_length = data_args.min_duration_in_seconds * feature_extractor.sampling_rate
522
+ max_labels_length = 448 # model.config.max_length
523
+
524
+ audio_column_name = data_args.audio_column_name
525
+ text_column_name = data_args.text_column_name
526
+ model_input_name = feature_extractor.model_input_names[0]
527
+ do_lower_case = data_args.do_lower_case
528
+ do_remove_punctuation = data_args.do_remove_punctuation
529
+ normalizer = BelarusianTextNormalizer() # custom normalizer based on 'official' text normalizer from OpenAI
530
+
531
+ if data_args.max_train_samples is not None:
532
+ raw_datasets["train"] = (
533
+ raw_datasets["train"].take(data_args.max_train_samples)
534
+ if data_args.streaming
535
+ else raw_datasets["train"].select(range(data_args.max_train_samples))
536
+ )
537
+
538
+ if data_args.max_eval_samples is not None:
539
+ raw_datasets["eval"] = (
540
+ raw_datasets["eval"].take(data_args.max_eval_samples)
541
+ if data_args.streaming
542
+ else raw_datasets["eval"].select(range(data_args.max_eval_samples))
543
+ )
544
+
545
+ def prepare_dataset(batch, labels_max_len: int = None):
546
+ # process audio
547
+ sample = batch[audio_column_name]
548
+ inputs = feature_extractor(sample["array"], sampling_rate=sample["sampling_rate"])
549
+ # process audio length
550
+ batch[model_input_name] = inputs.get(model_input_name)[0]
551
+ batch["input_length"] = len(sample["array"])
552
+
553
+ # process targets
554
+ input_str = batch[text_column_name].lower() if do_lower_case else batch[text_column_name]
555
+ if do_remove_punctuation:
556
+ input_str = normalizer(input_str).strip()
557
+ batch['labels'] = tokenizer(input_str).input_ids
558
+ batch['labels_length'] = len(batch['labels']) # include special characters
559
+
560
+ batch['labels_truncated'] = 0
561
+ # need to truncate validation and test labels that are longer that model.config.max_length.
562
+ # can't drop such examples because this will affect validation and test scores.
563
+ # thus need to truncate.
564
+ if labels_max_len is not None:
565
+ if len(batch['labels']) > labels_max_len:
566
+ batch['labels'] = batch['labels'][:labels_max_len]
567
+ batch['labels_truncated'] = 1
568
+
569
+ return batch
570
+
571
+ with training_args.main_process_first(desc="dataset map pre-processing"):
572
+ vectorized_datasets = IterableDatasetDict()
573
+
574
+ vectorized_datasets['train'] = raw_datasets['train'].map(
575
+ prepare_dataset, remove_columns=raw_datasets_features,
576
+ fn_kwargs=dict(labels_max_len=None),
577
+ ).with_format("torch")
578
+ vectorized_datasets['eval'] = raw_datasets['eval'].map(
579
+ prepare_dataset, remove_columns=raw_datasets_features,
580
+ fn_kwargs=dict(labels_max_len=max_labels_length),
581
+ ).with_format("torch")
582
+
583
+ if training_args.do_train and data_args.streaming:
584
+ # manually shuffle if streaming (done by the trainer for non-streaming)
585
+ vectorized_datasets["train"] = vectorized_datasets["train"].shuffle(
586
+ buffer_size=data_args.shuffle_buffer_size,
587
+ seed=training_args.seed,
588
+ )
589
+
590
+ # Filter training data that is shorter than min_input_length or longer than max_input_length.
591
+ # Drop items with labels longer that max model length.
592
+ # Drop such items from the train set only. Should keep them in eval set not to affect eval metrics.
593
+ def is_audio_in_length_range(length):
594
+ return min_input_length < length < max_input_length
595
+
596
+ def are_labels_in_length_range(labels_length):
597
+ return labels_length <= max_labels_length
598
+
599
+ if training_args.do_train:
600
+ # Filter items from train set only.
601
+ # Should keep them in eval set not to affect eval metrics.
602
+ vectorized_datasets["train"] = vectorized_datasets["train"].filter(
603
+ is_audio_in_length_range,
604
+ input_columns=["input_length"],
605
+ )
606
+ vectorized_datasets["train"] = vectorized_datasets["train"].filter(
607
+ are_labels_in_length_range,
608
+ input_columns=["labels_length"],
609
+ )
610
+
611
+ # 8. Load Metric
612
+ metric = evaluate.load("wer")
613
+ do_normalize_eval = data_args.do_normalize_eval
614
+
615
+ def compute_metrics(pred):
616
+ pred_ids = pred.predictions
617
+
618
+ pred.label_ids[pred.label_ids == -100] = tokenizer.pad_token_id
619
+
620
+ pred_str = tokenizer.batch_decode(pred_ids, skip_special_tokens=True)
621
+ # we do not want to group tokens when computing the metrics
622
+ label_str = tokenizer.batch_decode(pred.label_ids, skip_special_tokens=True)
623
+
624
+ if do_normalize_eval:
625
+ pred_str = [normalizer(pred) for pred in pred_str]
626
+ label_str = [normalizer(label) for label in label_str]
627
+ # filtering step to only evaluate the samples that correspond to non-zero references:
628
+ pred_str = [pred_str[i] for i in range(len(pred_str)) if len(label_str[i]) > 0]
629
+ label_str = [label_str[i] for i in range(len(label_str)) if len(label_str[i]) > 0]
630
+
631
+ wer = 100 * metric.compute(predictions=pred_str, references=label_str)
632
+
633
+ return {"wer": wer}
634
+
635
+ # 9. Create a single speech processor
636
+ if is_main_process(training_args.local_rank):
637
+ # save feature extractor, tokenizer and config
638
+ feature_extractor.save_pretrained(training_args.output_dir)
639
+ tokenizer.save_pretrained(training_args.output_dir)
640
+ config.save_pretrained(training_args.output_dir)
641
+
642
+ processor = AutoProcessor.from_pretrained(training_args.output_dir)
643
+
644
+ # 10. Define data collator
645
+ data_collator = DataCollatorSpeechSeq2SeqWithPadding(
646
+ processor=processor,
647
+ decoder_start_token_id=model.config.decoder_start_token_id,
648
+ )
649
+
650
+ # 11. Configure Trainer
651
+ # Trainer callback to reinitialise and reshuffle the streamable datasets at the beginning of each epoch
652
+ # Only required for streaming: Trainer automatically shuffles non-streaming datasets
653
+ class ShuffleCallback(TrainerCallback):
654
+ def on_epoch_begin(self, args, state, control, train_dataloader, **kwargs):
655
+ if isinstance(train_dataloader.dataset, IterableDatasetShard):
656
+ pass # set_epoch() is handled by the Trainer
657
+ elif isinstance(train_dataloader.dataset, IterableDataset):
658
+ train_dataloader.dataset.set_epoch(train_dataloader.dataset._epoch + 1)
659
+
660
+ # Initialize Trainer
661
+ trainer = Seq2SeqTrainer(
662
+ model=model,
663
+ args=training_args,
664
+ train_dataset=vectorized_datasets["train"] if training_args.do_train else None,
665
+ eval_dataset=vectorized_datasets["eval"] if training_args.do_eval else None,
666
+ tokenizer=processor,
667
+ data_collator=data_collator,
668
+ compute_metrics=compute_metrics if training_args.predict_with_generate else None,
669
+ callbacks=[ShuffleCallback()] if data_args.streaming else None,
670
+ )
671
+
672
+ # 12. Training
673
+ if training_args.do_train:
674
+ checkpoint = None
675
+ if training_args.resume_from_checkpoint is not None:
676
+ checkpoint = training_args.resume_from_checkpoint
677
+ elif last_checkpoint is not None:
678
+ checkpoint = last_checkpoint
679
+ train_result = trainer.train(resume_from_checkpoint=checkpoint)
680
+ trainer.save_model() # Saves the feature extractor too for easy upload
681
+
682
+ metrics = train_result.metrics
683
+ if data_args.max_train_samples:
684
+ metrics["train_samples"] = data_args.max_train_samples
685
+ trainer.log_metrics("train", metrics)
686
+ trainer.save_metrics("train", metrics)
687
+ trainer.save_state()
688
+
689
+ # 13. Evaluation
690
+ results = {}
691
+ if training_args.do_eval:
692
+ logger.info("*** Evaluate ***")
693
+ metrics = trainer.evaluate(
694
+ metric_key_prefix="eval",
695
+ max_length=training_args.generation_max_length,
696
+ num_beams=training_args.generation_num_beams,
697
+ )
698
+ if data_args.max_eval_samples:
699
+ metrics["eval_samples"] = data_args.max_eval_samples
700
+
701
+ trainer.log_metrics("eval", metrics)
702
+ trainer.save_metrics("eval", metrics)
703
+
704
+ # 14. Write Training Stats
705
+ kwargs = {
706
+ "finetuned_from": model_args.model_name_or_path,
707
+ "tasks": "automatic-speech-recognition",
708
+ "tags": "whisper-event",
709
+ }
710
+ if data_args.dataset_name is not None:
711
+ kwargs["dataset_tags"] = data_args.dataset_name
712
+ if data_args.dataset_config_name is not None:
713
+ kwargs["dataset"] = f"{data_args.dataset_name} {data_args.dataset_config_name}"
714
+ else:
715
+ kwargs["dataset"] = data_args.dataset_name
716
+ if "common_voice" in data_args.dataset_name:
717
+ kwargs["language"] = data_args.dataset_config_name[:2]
718
+ if model_args.model_index_name is not None:
719
+ kwargs["model_name"] = model_args.model_index_name
720
+
721
+ if training_args.push_to_hub:
722
+ trainer.push_to_hub(**kwargs)
723
+ else:
724
+ trainer.create_model_card(**kwargs)
725
+
726
+ return results
727
+
728
+
729
+ if __name__ == "__main__":
730
+ main()
src/setup_env.sh ADDED
@@ -0,0 +1,25 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ sudo add-apt-repository -y ppa:jonathonf/ffmpeg-4
2
+ sudo apt update
3
+ sudo apt install -y ffmpeg
4
+
5
+ sudo apt-get install git-lfs
6
+
7
+ sudo apt-get install tmux
8
+
9
+ cd ~
10
+ echo "executing env setup from $(pwd)"
11
+
12
+ python3 -m venv ~/python_venvs/hf_env
13
+ source ~/python_venvs/hf_env/bin/activate
14
+ echo "source ~/python_venvs/hf_env/bin/activate" >> ~/.bashrc
15
+
16
+ git clone https://github.com/yks72p/whisper-finetuning-be
17
+ pip install -r ~/whisper-finetuning-be/requirements.txt
18
+
19
+ git config --global credential.helper store
20
+ huggingface-cli login
21
+
22
+ echo "env setup"
23
+ echo "! PLEASE LOGIN INTO GIT TO BE ABLE TO PUSH TO HF HUB !"
24
+ echo "> git config --globase user.name <user_name>"
25
+ echo "> git config --globase user.email <user_email>"