esb
/

NeMo
English
esb
sanchit-gandhi HF staff commited on
Commit
320c039
1 Parent(s): 9c2e43c

Add training scripts and weights

Browse files
.gitattributes CHANGED
@@ -21,7 +21,6 @@
21
  *.pt filter=lfs diff=lfs merge=lfs -text
22
  *.pth filter=lfs diff=lfs merge=lfs -text
23
  *.rar filter=lfs diff=lfs merge=lfs -text
24
- *.safetensors filter=lfs diff=lfs merge=lfs -text
25
  saved_model/**/* filter=lfs diff=lfs merge=lfs -text
26
  *.tar.* filter=lfs diff=lfs merge=lfs -text
27
  *.tflite filter=lfs diff=lfs merge=lfs -text
@@ -31,3 +30,4 @@ saved_model/**/* filter=lfs diff=lfs merge=lfs -text
31
  *.zip filter=lfs diff=lfs merge=lfs -text
32
  *.zst filter=lfs diff=lfs merge=lfs -text
33
  *tfevents* filter=lfs diff=lfs merge=lfs -text
 
 
21
  *.pt filter=lfs diff=lfs merge=lfs -text
22
  *.pth filter=lfs diff=lfs merge=lfs -text
23
  *.rar filter=lfs diff=lfs merge=lfs -text
 
24
  saved_model/**/* filter=lfs diff=lfs merge=lfs -text
25
  *.tar.* filter=lfs diff=lfs merge=lfs -text
26
  *.tflite filter=lfs diff=lfs merge=lfs -text
 
30
  *.zip filter=lfs diff=lfs merge=lfs -text
31
  *.zst filter=lfs diff=lfs merge=lfs -text
32
  *tfevents* filter=lfs diff=lfs merge=lfs -text
33
+ *.nemo filter=lfs diff=lfs merge=lfs -text
README.md ADDED
@@ -0,0 +1,45 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ ---
2
+ language:
3
+ - en
4
+ tags:
5
+ - esb
6
+ datasets:
7
+ - esb/datasets
8
+ - kensho/spgispeech
9
+ ---
10
+ To reproduce this run, first install NVIDIA NeMo according to the [official instructions](https://github.com/NVIDIA/NeMo#installation), then execute:
11
+ ```python
12
+ #!/usr/bin/env bash
13
+ CUDA_VISIBLE_DEVICES=0 python run_speech_recognition_rnnt.py \
14
+ --config_path="conf/conformer_transducer_bpe_xlarge.yaml" \
15
+ --model_name_or_path="stt_en_conformer_transducer_xlarge" \
16
+ --dataset_name="esb/datasets" \
17
+ --tokenizer_path="tokenizer" \
18
+ --vocab_size="1024" \
19
+ --max_steps="100000" \
20
+ --dataset_config_name="spgispeech" \
21
+ --output_dir="./" \
22
+ --run_name="conformer-rnnt-spgispeech" \
23
+ --wandb_project="rnnt" \
24
+ --per_device_train_batch_size="8" \
25
+ --per_device_eval_batch_size="4" \
26
+ --logging_steps="50" \
27
+ --learning_rate="1e-4" \
28
+ --warmup_steps="500" \
29
+ --save_strategy="steps" \
30
+ --save_steps="20000" \
31
+ --evaluation_strategy="steps" \
32
+ --eval_steps="20000" \
33
+ --report_to="wandb" \
34
+ --preprocessing_num_workers="4" \
35
+ --fused_batch_size="4" \
36
+ --length_column_name="input_lengths" \
37
+ --fuse_loss_wer \
38
+ --group_by_length \
39
+ --overwrite_output_dir \
40
+ --do_train \
41
+ --do_eval \
42
+ --do_predict \
43
+ --use_auth_token
44
+
45
+ ```
conf/conformer_transducer_bpe_xlarge.yaml ADDED
File without changes
models/__init__.py ADDED
@@ -0,0 +1 @@
 
 
1
+ from .modeling_rnnt import RNNTBPEModel
models/modeling_rnnt.py ADDED
@@ -0,0 +1,126 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ from dataclasses import dataclass
2
+ from typing import Optional
3
+
4
+ import torch
5
+ from nemo.collections.asr.models import EncDecRNNTBPEModel
6
+ from omegaconf import DictConfig
7
+ from transformers.utils import ModelOutput
8
+
9
+
10
+ @dataclass
11
+ class RNNTOutput(ModelOutput):
12
+ """
13
+ Base class for RNNT outputs.
14
+ """
15
+
16
+ loss: Optional[torch.FloatTensor] = None
17
+ wer: Optional[float] = None
18
+ wer_num: Optional[float] = None
19
+ wer_denom: Optional[float] = None
20
+
21
+
22
+ # Adapted from https://github.com/NVIDIA/NeMo/blob/66c7677cd4a68d78965d4905dd1febbf5385dff3/nemo/collections/asr/models/rnnt_bpe_models.py#L33
23
+ class RNNTBPEModel(EncDecRNNTBPEModel):
24
+ def __init__(self, cfg: DictConfig):
25
+ super().__init__(cfg=cfg, trainer=None)
26
+
27
+ def encoding(
28
+ self, input_signal=None, input_signal_length=None, processed_signal=None, processed_signal_length=None
29
+ ):
30
+ """
31
+ Forward pass of the acoustic model. Note that for RNNT Models, the forward pass of the model is a 3 step process,
32
+ and this method only performs the first step - forward of the acoustic model.
33
+
34
+ Please refer to the `forward` in order to see the full `forward` step for training - which
35
+ performs the forward of the acoustic model, the prediction network and then the joint network.
36
+ Finally, it computes the loss and possibly compute the detokenized text via the `decoding` step.
37
+
38
+ Please refer to the `validation_step` in order to see the full `forward` step for inference - which
39
+ performs the forward of the acoustic model, the prediction network and then the joint network.
40
+ Finally, it computes the decoded tokens via the `decoding` step and possibly compute the batch metrics.
41
+
42
+ Args:
43
+ input_signal: Tensor that represents a batch of raw audio signals,
44
+ of shape [B, T]. T here represents timesteps, with 1 second of audio represented as
45
+ `self.sample_rate` number of floating point values.
46
+ input_signal_length: Vector of length B, that contains the individual lengths of the audio
47
+ sequences.
48
+ processed_signal: Tensor that represents a batch of processed audio signals,
49
+ of shape (B, D, T) that has undergone processing via some DALI preprocessor.
50
+ processed_signal_length: Vector of length B, that contains the individual lengths of the
51
+ processed audio sequences.
52
+
53
+ Returns:
54
+ A tuple of 2 elements -
55
+ 1) The log probabilities tensor of shape [B, T, D].
56
+ 2) The lengths of the acoustic sequence after propagation through the encoder, of shape [B].
57
+ """
58
+ has_input_signal = input_signal is not None and input_signal_length is not None
59
+ has_processed_signal = processed_signal is not None and processed_signal_length is not None
60
+ if (has_input_signal ^ has_processed_signal) is False:
61
+ raise ValueError(
62
+ f"{self} Arguments ``input_signal`` and ``input_signal_length`` are mutually exclusive "
63
+ " with ``processed_signal`` and ``processed_signal_len`` arguments."
64
+ )
65
+
66
+ if not has_processed_signal:
67
+ processed_signal, processed_signal_length = self.preprocessor(
68
+ input_signal=input_signal, length=input_signal_length,
69
+ )
70
+
71
+ # Spec augment is not applied during evaluation/testing
72
+ if self.spec_augmentation is not None and self.training:
73
+ processed_signal = self.spec_augmentation(input_spec=processed_signal, length=processed_signal_length)
74
+
75
+ encoded, encoded_len = self.encoder(audio_signal=processed_signal, length=processed_signal_length)
76
+ return encoded, encoded_len
77
+
78
+ def forward(self, input_ids, input_lengths=None, labels=None, label_lengths=None):
79
+ # encoding() only performs encoder forward
80
+ encoded, encoded_len = self.encoding(input_signal=input_ids, input_signal_length=input_lengths)
81
+ del input_ids
82
+
83
+ # During training, loss must be computed, so decoder forward is necessary
84
+ decoder, target_length, states = self.decoder(targets=labels, target_length=label_lengths)
85
+
86
+ # If experimental fused Joint-Loss-WER is not used
87
+ if not self.joint.fuse_loss_wer:
88
+ # Compute full joint and loss
89
+ joint = self.joint(encoder_outputs=encoded, decoder_outputs=decoder)
90
+ loss_value = self.loss(
91
+ log_probs=joint, targets=labels, input_lengths=encoded_len, target_lengths=target_length
92
+ )
93
+ # Add auxiliary losses, if registered
94
+ loss_value = self.add_auxiliary_losses(loss_value)
95
+ wer = wer_num = wer_denom = None
96
+ if not self.training:
97
+ self.wer.update(encoded, encoded_len, labels, target_length)
98
+ wer, wer_num, wer_denom = self.wer.compute()
99
+ self.wer.reset()
100
+
101
+ else:
102
+ # If experimental fused Joint-Loss-WER is used
103
+ # Fused joint step
104
+ loss_value, wer, wer_num, wer_denom = self.joint(
105
+ encoder_outputs=encoded,
106
+ decoder_outputs=decoder,
107
+ encoder_lengths=encoded_len,
108
+ transcripts=labels,
109
+ transcript_lengths=label_lengths,
110
+ compute_wer=not self.training,
111
+ )
112
+ # Add auxiliary losses, if registered
113
+ loss_value = self.add_auxiliary_losses(loss_value)
114
+
115
+ return RNNTOutput(loss=loss_value, wer=wer, wer_num=wer_num, wer_denom=wer_denom)
116
+
117
+ def transcribe(self, input_ids, input_lengths=None, labels=None, label_lengths=None, return_hypotheses: bool = False, partial_hypothesis: Optional = None):
118
+ encoded, encoded_len = self.encoding(input_signal=input_ids, input_signal_length=input_lengths)
119
+ del input_ids
120
+ best_hyp, all_hyp = self.decoding.rnnt_decoder_predictions_tensor(
121
+ encoded,
122
+ encoded_len,
123
+ return_hypotheses=return_hypotheses,
124
+ partial_hypotheses=partial_hypothesis,
125
+ )
126
+ return best_hyp, all_hyp
run_speech_recognition_rnnt.py ADDED
@@ -0,0 +1,789 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ #!/usr/bin/env python
2
+ # coding=utf-8
3
+ #
4
+ # Licensed under the Apache License, Version 2.0 (the "License");
5
+ # you may not use this file except in compliance with the License.
6
+ # You may obtain a copy of the License at
7
+ #
8
+ # http://www.apache.org/licenses/LICENSE-2.0
9
+ #
10
+ # Unless required by applicable law or agreed to in writing, software
11
+ # distributed under the License is distributed on an "AS IS" BASIS,
12
+ # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13
+ # See the License for the specific language governing permissions and
14
+ # limitations under the License.
15
+ """
16
+ Fine-tuning NVIDIA RNN-T models for speech recognition.
17
+ """
18
+ # You can also adapt this script on your own sequence to sequence task. Pointers for this are left as comments.
19
+ import copy
20
+ import logging
21
+ import os
22
+ import sys
23
+ from dataclasses import dataclass, field
24
+
25
+ import wandb
26
+ from torch.utils.data import Dataset
27
+ from tqdm import tqdm
28
+ import json
29
+ from typing import Optional, Dict, Union, List, Any
30
+
31
+ import numpy as np
32
+ import torch
33
+
34
+ from omegaconf import OmegaConf
35
+ from models import RNNTBPEModel
36
+
37
+ import datasets
38
+ from datasets import DatasetDict, load_dataset, load_metric
39
+ import transformers
40
+ from transformers import (
41
+ HfArgumentParser,
42
+ Seq2SeqTrainingArguments,
43
+ set_seed,
44
+ Trainer,
45
+ )
46
+ from transformers.trainer_utils import get_last_checkpoint, is_main_process
47
+ from transformers.utils import check_min_version
48
+ from transformers.utils.versions import require_version
49
+
50
+ from process_asr_text_tokenizer import __process_data as nemo_process_data, \
51
+ __build_document_from_manifests as nemo_build_document_from_manifests
52
+
53
+
54
+ # Will error if the minimal version of Transformers is not installed. Remove at your own risks.
55
+ check_min_version("4.17.0.dev0")
56
+
57
+ require_version("datasets>=1.18.0", "To fix: pip install -r examples/pytorch/speech-recognition/requirements.txt")
58
+
59
+ logger = logging.getLogger(__name__)
60
+
61
+
62
+ @dataclass
63
+ class ModelArguments:
64
+ """
65
+ Arguments pertaining to which model/config/tokenizer we are going to fine-tune from.
66
+ """
67
+
68
+ config_path: str = field(
69
+ metadata={"help": "Path to pretrained model or model identifier from huggingface.co/models."},
70
+ )
71
+ model_name_or_path: Optional[str] = field(
72
+ default=None,
73
+ metadata={"help": "Path to pretrained model or model identifier from NVIDIA NeMo NGC."}
74
+ )
75
+ pretrained_model_name_or_path: Optional[str] = field(
76
+ default=None,
77
+ metadata={"help": "Path to local pretrained model or model identifier."}
78
+ )
79
+ cache_dir: Optional[str] = field(
80
+ default=None,
81
+ metadata={"help": "Where to store the pretrained models downloaded from huggingface.co or NVIDIA NeMo NGC."},
82
+ )
83
+ use_auth_token: bool = field(
84
+ default=False,
85
+ metadata={
86
+ "help": "Will use the token generated when running `transformers-cli login` (necessary to use this script "
87
+ "with private models)."
88
+ },
89
+ )
90
+ manifest_path: str = field(
91
+ default="data",
92
+ metadata={
93
+ "help": "Manifest path."
94
+ },
95
+ )
96
+ tokenizer_path: str = field(
97
+ default="tokenizers",
98
+ metadata={
99
+ "help": "Tokenizer path."
100
+ },
101
+ )
102
+ vocab_size: int = field(
103
+ default=1024,
104
+ metadata={"help": "Tokenizer vocab size."}
105
+ )
106
+ tokenizer_type: str = field(
107
+ default="spe",
108
+ metadata={
109
+ "help": "Can be either spe or wpe. spe refers to the Google sentencepiece library tokenizer."
110
+ "wpe refers to the HuggingFace BERT Word Piece tokenizer."
111
+ },
112
+ )
113
+ spe_type: str = field(
114
+ default="bpe",
115
+ metadata={
116
+ "help": "Type of the SentencePiece model. Can be `bpe`, `unigram`, `char` or `word`."
117
+ "Used only if `tokenizer_type` == `spe`"
118
+ },
119
+ )
120
+ cutoff_freq: str = field(
121
+ default=0.001,
122
+ metadata={"help": "Drop the least frequent chars from the train set when building the tokenizer."}
123
+ )
124
+ fuse_loss_wer: bool = field(
125
+ default=True,
126
+ metadata={
127
+ "help": "Whether to fuse the computation of prediction net + joint net + loss + WER calculation to be run "
128
+ "on sub-batches of size `fused_batch_size`"
129
+ }
130
+ )
131
+ fused_batch_size: int = field(
132
+ default=8,
133
+ metadata={
134
+ "help": "`fused_batch_size` is the actual batch size of the prediction net, joint net and transducer loss."
135
+ "Using small values here will preserve a lot of memory during training, but will make training slower as well."
136
+ "An optimal ratio of fused_batch_size : per_device_train_batch_size is 1:1."
137
+ "However, to preserve memory, this ratio can be 1:8 or even 1:16."
138
+ }
139
+ )
140
+ final_decoding_strategy: str = field(
141
+ default="greedy_batch",
142
+ metadata={
143
+ "help": "Decoding strategy for final eval/prediction steps. One of: [`greedy`, `greedy_batch`, `beam`, "
144
+ "`tsd`, `alsd`]."
145
+ }
146
+ )
147
+ final_num_beams: int = field(
148
+ default=1,
149
+ metadata={
150
+ "help": "Number of beams for final eval/prediction steps. Increase beam size for better scores, "
151
+ "but it will take much longer for transcription!"
152
+ }
153
+ )
154
+
155
+
156
+ @dataclass
157
+ class DataTrainingArguments:
158
+ """
159
+ Arguments pertaining to what data we are going to input our model for training and eval.
160
+ """
161
+
162
+ dataset_name: str = field(
163
+ default=None, metadata={"help": "The name of the dataset to use (via the datasets library)."}
164
+ )
165
+ dataset_config_name: Optional[str] = field(
166
+ default=None, metadata={"help": "The configuration name of the dataset to use (via the datasets library)."}
167
+ )
168
+ text_column: Optional[str] = field(
169
+ default=None,
170
+ metadata={"help": "The name of the column in the datasets containing the full texts (for summarization)."},
171
+ )
172
+ dataset_cache_dir: Optional[str] = field(
173
+ default=None, metadata={"help": "Path to cache directory for saving and loading datasets"}
174
+ )
175
+ overwrite_cache: bool = field(
176
+ default=False, metadata={"help": "Overwrite the cached training and evaluation sets"}
177
+ )
178
+ preprocessing_num_workers: Optional[int] = field(
179
+ default=None,
180
+ metadata={"help": "The number of processes to use for the preprocessing."},
181
+ )
182
+ max_train_samples: Optional[int] = field(
183
+ default=None,
184
+ metadata={
185
+ "help": "For debugging purposes or quicker training, truncate the number of training examples to this "
186
+ "value if set."
187
+ },
188
+ )
189
+ max_eval_samples: Optional[int] = field(
190
+ default=None,
191
+ metadata={
192
+ "help": "For debugging purposes or quicker training, truncate the number of evaluation examples to this "
193
+ "value if set."
194
+ },
195
+ )
196
+ max_predict_samples: Optional[int] = field(
197
+ default=None,
198
+ metadata={
199
+ "help": "For debugging purposes or quicker training, truncate the number of test examples to this "
200
+ "value if set."
201
+ },
202
+ )
203
+ audio_column_name: str = field(
204
+ default="audio",
205
+ metadata={"help": "The name of the dataset column containing the audio data. Defaults to 'audio'"},
206
+ )
207
+ text_column_name: str = field(
208
+ default="text",
209
+ metadata={"help": "The name of the dataset column containing the text data. Defaults to 'text'"},
210
+ )
211
+ max_duration_in_seconds: float = field(
212
+ default=20.0,
213
+ metadata={
214
+ "help": "Truncate training audio files that are longer than `max_duration_in_seconds` seconds to 'max_duration_in_seconds`"
215
+ },
216
+ )
217
+ min_duration_in_seconds: float = field(
218
+ default=0.0, metadata={"help": "Filter audio files that are shorter than `min_duration_in_seconds` seconds"}
219
+ )
220
+ max_eval_duration_in_seconds: float = field(
221
+ default=None,
222
+ metadata={
223
+ "help": "Truncate eval/test audio files that are longer than `max_duration_in_seconds` seconds to 'max_duration_in_seconds`"
224
+ },
225
+ )
226
+ max_target_length: Optional[int] = field(
227
+ default=128,
228
+ metadata={
229
+ "help": "The maximum total sequence length for target text after tokenization. Sequences longer "
230
+ "than this will be truncated, sequences shorter will be padded."
231
+ },
232
+ )
233
+ min_target_length: Optional[int] = field(
234
+ default=2,
235
+ metadata={
236
+ "help": "The minimum total sequence length for target text after tokenization. Sequences shorter "
237
+ "than this will be filtered."
238
+ },
239
+ )
240
+ preprocessing_only: bool = field(
241
+ default=False,
242
+ metadata={
243
+ "help": "Whether to only do data preprocessing and skip training. "
244
+ "This is especially useful when data preprocessing errors out in distributed training due to timeout. "
245
+ "In this case, one should run the preprocessing in a non-distributed setup with `preprocessing_only=True` "
246
+ "so that the cached datasets can consequently be loaded in distributed training"
247
+ },
248
+ )
249
+ train_split_name: str = field(
250
+ default="train",
251
+ metadata={
252
+ "help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'"
253
+ },
254
+ )
255
+ eval_split_name: str = field(
256
+ default="validation",
257
+ metadata={
258
+ "help": "The name of the evaluation data set split to use (via the datasets library). Defaults to 'validation'"
259
+ },
260
+ )
261
+ test_split_name: str = field(
262
+ default="test",
263
+ metadata={"help": "The name of the test data set split to use (via the datasets library). Defaults to 'test'"},
264
+ )
265
+ do_lower_case: bool = field(
266
+ default=True,
267
+ metadata={"help": "Whether the target text should be lower cased."},
268
+ )
269
+ wandb_project: str = field(
270
+ default="speech-recognition-rnnt",
271
+ metadata={"help": "The name of the wandb project."},
272
+ )
273
+
274
+
275
+ def write_wandb_pred(pred_str, label_str, prefix="eval"):
276
+ # convert str data to a wandb compatible format
277
+ str_data = [[label_str[i], pred_str[i]] for i in range(len(pred_str))]
278
+ # we'll log all predictions for the last epoch
279
+ wandb.log(
280
+ {
281
+ f"{prefix}/predictions": wandb.Table(
282
+ columns=["label_str", "pred_str"], data=str_data
283
+ )
284
+ },
285
+ )
286
+
287
+
288
+ def build_tokenizer(model_args, data_args, manifests):
289
+ """
290
+ Function to build a NeMo tokenizer from manifest file(s).
291
+ Copied from https://github.com/NVIDIA/NeMo/blob/66c7677cd4a68d78965d4905dd1febbf5385dff3/scripts/tokenizers/process_asr_text_tokenizer.py#L268
292
+ """
293
+ data_root = model_args.tokenizer_path
294
+ if isinstance(manifests, list):
295
+ joint_manifests = ",".join(manifests)
296
+ else:
297
+ joint_manifests = manifests
298
+ vocab_size = model_args.vocab_size
299
+ tokenizer = model_args.tokenizer_type
300
+ spe_type = model_args.spe_type
301
+ if not 0 <= model_args.cutoff_freq < 1:
302
+ raise ValueError(f"`cutoff_freq` must be between zero and one, got {model_args.cutoff_freq}")
303
+ spe_character_coverage = 1 - model_args.cutoff_freq
304
+
305
+ logger.info("Building tokenizer...")
306
+ if not os.path.exists(data_root):
307
+ os.makedirs(data_root)
308
+
309
+ text_corpus_path = nemo_build_document_from_manifests(data_root, joint_manifests)
310
+
311
+ tokenizer_path = nemo_process_data(
312
+ text_corpus_path,
313
+ data_root,
314
+ vocab_size,
315
+ tokenizer,
316
+ spe_type,
317
+ lower_case=data_args.do_lower_case,
318
+ spe_character_coverage=spe_character_coverage,
319
+ spe_sample_size=-1,
320
+ spe_train_extremely_large_corpus=False,
321
+ spe_max_sentencepiece_length=-1,
322
+ spe_bos=False,
323
+ spe_eos=False,
324
+ spe_pad=False,
325
+ )
326
+
327
+ print("Serialized tokenizer at location :", tokenizer_path)
328
+ logger.info('Done!')
329
+
330
+ # Tokenizer path
331
+ if tokenizer == 'spe':
332
+ tokenizer_dir = os.path.join(data_root, f"tokenizer_spe_{spe_type}_v{vocab_size}")
333
+ tokenizer_type_cfg = "bpe"
334
+ else:
335
+ tokenizer_dir = os.path.join(data_root, f"tokenizer_wpe_v{vocab_size}")
336
+ tokenizer_type_cfg = "wpe"
337
+
338
+ return tokenizer_dir, tokenizer_type_cfg
339
+
340
+
341
+ def NeMoDataCollator(features: List[Dict[str, Union[List[int], torch.Tensor]]]) -> Dict[str, torch.Tensor]:
342
+ """
343
+ Data collator that will dynamically pad the inputs received.
344
+ Since NeMo models don't have a HF processor defined (feature extractor + tokenizer), we'll pad by hand...
345
+ The padding idx is arbitrary: we provide the model with the input lengths and label lengths, from which
346
+ all the relevant padding information is inferred. Thus, we'll use the default np.pad padding idx (0).
347
+ """
348
+ # split inputs and labels since they have to be of different lengths
349
+ # and need different padding methods
350
+ input_ids = [feature["input_ids"] for feature in features]
351
+ labels = [feature["labels"] for feature in features]
352
+
353
+ # first, pad the audio inputs to max_len
354
+ input_lengths = [feature["input_lengths"] for feature in features]
355
+ max_input_len = max(input_lengths)
356
+ input_ids = [np.pad(input_val, (0, max_input_len - input_len), 'constant') for input_val, input_len in
357
+ zip(input_ids, input_lengths)]
358
+
359
+ # next, pad the target labels to max_len
360
+ label_lengths = [len(lab) for lab in labels]
361
+ max_label_len = max(label_lengths)
362
+ labels = [np.pad(lab, (0, max_label_len - lab_len), 'constant') for lab, lab_len in zip(labels, label_lengths)]
363
+
364
+ batch = {"input_lengths": input_lengths, "labels": labels, "label_lengths": label_lengths}
365
+
366
+ # return batch as a pt tensor (list -> np.array -> torch.tensor)
367
+ batch = {k: torch.tensor(np.array(v), requires_grad=False) for k, v in batch.items()}
368
+
369
+ # leave all ints as are, convert float64 to pt float
370
+ batch["input_ids"] = torch.tensor(np.array(input_ids, dtype=np.float32), requires_grad=False)
371
+
372
+ return batch
373
+
374
+
375
+ def main():
376
+ # See all possible arguments in src/transformers/training_args.py
377
+ # or by passing the --help flag to this script.
378
+ # We now keep distinct sets of args, for a cleaner separation of concerns.
379
+
380
+ parser = HfArgumentParser((ModelArguments, DataTrainingArguments, Seq2SeqTrainingArguments))
381
+ if len(sys.argv) == 2 and sys.argv[1].endswith(".json"):
382
+ # If we pass only one argument to the script and it's the path to a json file,
383
+ # let's parse it to get our arguments.
384
+ model_args, data_args, training_args = parser.parse_json_file(json_file=os.path.abspath(sys.argv[1]))
385
+ else:
386
+ model_args, data_args, training_args = parser.parse_args_into_dataclasses()
387
+
388
+ # Set wandb project ID before instantiating the Trainer
389
+ os.environ["WANDB_PROJECT"] = data_args.wandb_project
390
+
391
+ # Detecting last checkpoint.
392
+ last_checkpoint = None
393
+ if os.path.isdir(training_args.output_dir) and training_args.do_train and not training_args.overwrite_output_dir:
394
+ last_checkpoint = get_last_checkpoint(training_args.output_dir)
395
+ if last_checkpoint is None and len(os.listdir(training_args.output_dir)) > 0:
396
+ raise ValueError(
397
+ f"Output directory ({training_args.output_dir}) already exists and is not empty. "
398
+ "Use --overwrite_output_dir to overcome."
399
+ )
400
+ elif last_checkpoint is not None:
401
+ logger.info(
402
+ f"Checkpoint detected, resuming training at {last_checkpoint}. To avoid this behavior, change "
403
+ "the `--output_dir` or add `--overwrite_output_dir` to train from scratch."
404
+ )
405
+
406
+ # Setup logging
407
+ logging.basicConfig(
408
+ format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
409
+ datefmt="%m/%d/%Y %H:%M:%S",
410
+ handlers=[logging.StreamHandler(sys.stdout)],
411
+ )
412
+ logger.setLevel(logging.INFO if is_main_process(training_args.local_rank) else logging.WARN)
413
+
414
+ # Log on each process the small summary:
415
+ logger.warning(
416
+ f"Process rank: {training_args.local_rank}, device: {training_args.device}, n_gpu: {training_args.n_gpu}"
417
+ f"distributed training: {bool(training_args.local_rank != -1)}, 16-bits training: {training_args.fp16}"
418
+ )
419
+ # Set the verbosity to info of the Transformers logger (on main process only):
420
+ if is_main_process(training_args.local_rank):
421
+ transformers.utils.logging.set_verbosity_info()
422
+ logger.info("Training/evaluation parameters %s", training_args)
423
+
424
+ # Set seed before initializing model.
425
+ set_seed(training_args.seed)
426
+
427
+ # load the model config (discarding optimiser and trainer attributes)
428
+ config = OmegaConf.load(model_args.config_path).model
429
+
430
+ # 4. Load dataset
431
+ raw_datasets = DatasetDict()
432
+
433
+ if training_args.do_train:
434
+ raw_datasets["train"] = load_dataset(
435
+ data_args.dataset_name,
436
+ data_args.dataset_config_name,
437
+ split=data_args.train_split_name,
438
+ cache_dir=data_args.dataset_cache_dir,
439
+ use_auth_token=True if model_args.use_auth_token else None,
440
+ )
441
+
442
+ if training_args.do_eval:
443
+ raw_datasets["eval"] = load_dataset(
444
+ data_args.dataset_name,
445
+ data_args.dataset_config_name,
446
+ split=data_args.eval_split_name,
447
+ cache_dir=data_args.dataset_cache_dir,
448
+ use_auth_token=True if model_args.use_auth_token else None,
449
+ )
450
+
451
+ if training_args.do_predict:
452
+ test_split = data_args.test_split_name.split("+")
453
+ for split in test_split:
454
+ raw_datasets[split] = load_dataset(
455
+ data_args.dataset_name,
456
+ data_args.dataset_config_name,
457
+ split=split,
458
+ cache_dir=data_args.dataset_cache_dir,
459
+ use_auth_token=True if model_args.use_auth_token else None,
460
+ )
461
+
462
+ if not training_args.do_train and not training_args.do_eval and not training_args.do_predict:
463
+ raise ValueError(
464
+ "Cannot not train, not do evaluation and not do prediction. At least one of "
465
+ "training, evaluation or prediction has to be done."
466
+ )
467
+
468
+ # if not training, there is no need to run multiple epochs
469
+ if not training_args.do_train:
470
+ training_args.num_train_epochs = 1
471
+
472
+ if data_args.audio_column_name not in next(iter(raw_datasets.values())).column_names:
473
+ raise ValueError(
474
+ f"--audio_column_name '{data_args.audio_column_name}' not found in dataset '{data_args.dataset_name}'. "
475
+ "Make sure to set `--audio_column_name` to the correct audio column - one of "
476
+ f"{', '.join(next(iter(raw_datasets.values())).column_names)}."
477
+ )
478
+
479
+ if data_args.text_column_name not in next(iter(raw_datasets.values())).column_names:
480
+ raise ValueError(
481
+ f"--text_column_name {data_args.text_column_name} not found in dataset '{data_args.dataset_name}'. "
482
+ "Make sure to set `--text_column_name` to the correct text column - one of "
483
+ f"{', '.join(next(iter(raw_datasets.values())).column_names)}."
484
+ )
485
+
486
+ # 6. Resample speech dataset ALWAYS
487
+ raw_datasets = raw_datasets.cast_column(
488
+ data_args.audio_column_name, datasets.features.Audio(sampling_rate=config.sample_rate)
489
+ )
490
+
491
+ # 7. Preprocessing the datasets.
492
+ # We need to read the audio files as arrays and tokenize the targets.
493
+ max_input_length = int(data_args.max_duration_in_seconds * config.sample_rate)
494
+ min_input_length = max(int(data_args.min_duration_in_seconds * config.sample_rate), 1)
495
+ max_eval_input_length = int(data_args.max_eval_duration_in_seconds * config.sample_rate) if data_args.max_eval_duration_in_seconds else None
496
+ audio_column_name = data_args.audio_column_name
497
+ num_workers = data_args.preprocessing_num_workers
498
+ text_column_name = data_args.text_column_name
499
+
500
+ if training_args.do_train and data_args.max_train_samples is not None:
501
+ raw_datasets["train"] = raw_datasets["train"].select(range(data_args.max_train_samples))
502
+
503
+ if training_args.do_eval and data_args.max_eval_samples is not None:
504
+ raw_datasets["eval"] = raw_datasets["eval"].select(range(data_args.max_eval_samples))
505
+
506
+ if training_args.do_predict and data_args.max_predict_samples is not None:
507
+ for split in test_split:
508
+ raw_datasets[split] = raw_datasets[split].select(range(data_args.max_predict_samples))
509
+
510
+ # Function to build a NeMo tokenizer manifest from a HF dataset
511
+ # TODO: with a bit of hacking around we can probably bypass this step entirely
512
+ def build_manifest(ds, manifest_path):
513
+ with open(manifest_path, 'w') as fout:
514
+ for sample in tqdm(ds[text_column_name]):
515
+ # Write the metadata to the manifest
516
+ metadata = {
517
+ "text": sample
518
+ }
519
+ json.dump(metadata, fout)
520
+ fout.write('\n')
521
+
522
+ config.train_ds = config.validation_ds = config.test_ds = None
523
+
524
+ if not os.path.exists(model_args.manifest_path) and training_args.do_train:
525
+ os.makedirs(model_args.manifest_path)
526
+ manifest = os.path.join(model_args.manifest_path, "train.json")
527
+ logger.info(f"Building training manifest at {manifest}")
528
+ build_manifest(raw_datasets["train"], manifest)
529
+ else:
530
+ manifest = os.path.join(model_args.manifest_path, "train.json")
531
+ logger.info(f"Re-using training manifest at {manifest}")
532
+
533
+ tokenizer_dir, tokenizer_type_cfg = build_tokenizer(model_args, data_args, manifest)
534
+
535
+ # generalise the script later to load a pre-built tokenizer for eval only
536
+ config.tokenizer.dir = tokenizer_dir
537
+ config.tokenizer.type = tokenizer_type_cfg
538
+
539
+ # possibly fused-computation of prediction net + joint net + loss + WER calculation
540
+ config.joint.fuse_loss_wer = model_args.fuse_loss_wer
541
+ if model_args.fuse_loss_wer:
542
+ config.joint.fused_batch_size = model_args.fused_batch_size
543
+
544
+ if model_args.model_name_or_path is not None:
545
+ # load pre-trained model weights
546
+ model = RNNTBPEModel.from_pretrained(model_args.model_name_or_path, override_config_path=config,
547
+ map_location="cpu")
548
+ model.save_name = model_args.model_name_or_path
549
+
550
+ pretrained_decoder = model.decoder.state_dict()
551
+ pretrained_joint = model.joint.state_dict()
552
+ model.change_vocabulary(new_tokenizer_dir=tokenizer_dir, new_tokenizer_type=tokenizer_type_cfg)
553
+
554
+ # TODO: add checks for loading decoder/joint state dict
555
+ model.decoder.load_state_dict(pretrained_decoder)
556
+ model.joint.load_state_dict(pretrained_joint)
557
+
558
+ elif model_args.pretrained_model_name_or_path is not None:
559
+ model = RNNTBPEModel.restore_from(model_args.pretrained_model_name_or_path, override_config_path=config,
560
+ map_location="cpu")
561
+ model.save_name = model_args.config_path.split("/")[-1].split(".")[0]
562
+
563
+ else:
564
+ model = RNNTBPEModel(cfg=config)
565
+ model.save_name = model_args.config_path.split("/")[-1].split(".")[0]
566
+ model.change_vocabulary(new_tokenizer_dir=tokenizer_dir, new_tokenizer_type=tokenizer_type_cfg)
567
+
568
+ # now that we have our model and tokenizer defined, we can tokenize the text data
569
+ tokenizer = model.tokenizer.tokenizer.encode_as_ids
570
+
571
+ def prepare_dataset(batch):
572
+ # pre-process audio
573
+ sample = batch[audio_column_name]
574
+
575
+ # NeMo RNNT model performs the audio preprocessing in the `.forward()` call
576
+ # => we only need to supply it with the raw audio values
577
+ batch["input_ids"] = sample["array"]
578
+ batch["input_lengths"] = len(sample["array"])
579
+
580
+ batch["labels"] = tokenizer(batch[text_column_name])
581
+ return batch
582
+
583
+ vectorized_datasets = raw_datasets.map(
584
+ prepare_dataset,
585
+ remove_columns=next(iter(raw_datasets.values())).column_names,
586
+ num_proc=num_workers,
587
+ desc="preprocess train dataset",
588
+ )
589
+
590
+ # filter training data with inputs shorter than min_input_length or longer than max_input_length
591
+ def is_audio_in_length_range(length):
592
+ return min_input_length < length < max_input_length
593
+
594
+ vectorized_datasets = vectorized_datasets.filter(
595
+ is_audio_in_length_range,
596
+ num_proc=num_workers,
597
+ input_columns=["input_lengths"],
598
+ )
599
+
600
+ if max_eval_input_length is not None:
601
+ # filter training data with inputs longer than max_input_length
602
+ def is_eval_audio_in_length_range(length):
603
+ return min_input_length < length < max_eval_input_length
604
+
605
+ vectorized_datasets = vectorized_datasets.filter(
606
+ is_eval_audio_in_length_range,
607
+ num_proc=num_workers,
608
+ input_columns=["input_lengths"],
609
+ )
610
+
611
+ # for large datasets it is advised to run the preprocessing on a
612
+ # single machine first with `args.preprocessing_only` since there will mostly likely
613
+ # be a timeout when running the script in distributed mode.
614
+ # In a second step `args.preprocessing_only` can then be set to `False` to load the
615
+ # cached dataset
616
+ if data_args.preprocessing_only:
617
+ cache = {k: v.cache_files for k, v in vectorized_datasets.items()}
618
+ logger.info(f"Data preprocessing finished. Files cached at {cache}.")
619
+ return
620
+
621
+
622
+ def compute_metrics(pred):
623
+ # Tuple of WERs returned by the model during eval: (wer, wer_num, wer_denom)
624
+ wer_num = pred.predictions[1]
625
+ wer_denom = pred.predictions[2]
626
+ # compute WERs over concat batches
627
+ wer = sum(wer_num) / sum(wer_denom)
628
+ return {"wer": wer}
629
+
630
+
631
+ class NeMoTrainer(Trainer):
632
+ def _save(self, output_dir: Optional[str] = None, state_dict=None):
633
+ # If we are executing this function, we are the process zero, so we don't check for that.
634
+ output_dir = output_dir if output_dir is not None else self.args.output_dir
635
+ os.makedirs(output_dir, exist_ok=True)
636
+ logger.info(f"Saving model checkpoint to {output_dir}")
637
+ # Save a trained model and configuration using `save_pretrained()`.
638
+ # They can then be reloaded using `from_pretrained()`
639
+ self.model.save_to(save_path=os.path.join(output_dir, model.save_name + ".nemo"))
640
+ # Good practice: save your training arguments together with the trained model
641
+ torch.save(self.args, os.path.join(output_dir, "training_args.bin"))
642
+
643
+ def transcribe(self, test_dataset: Dataset) -> List[Any]:
644
+ self.model.eval()
645
+ test_dataloader = self.get_test_dataloader(test_dataset)
646
+ hypotheses = []
647
+ for test_batch in tqdm(test_dataloader, desc="Transcribing"):
648
+ inputs = self._prepare_inputs(test_batch)
649
+ best_hyp, all_hyp = self.model.transcribe(**inputs)
650
+ hypotheses += best_hyp
651
+ del test_batch
652
+ return hypotheses
653
+
654
+
655
+ # Initialize Trainer
656
+ trainer = NeMoTrainer(
657
+ model=model,
658
+ args=training_args,
659
+ compute_metrics=compute_metrics,
660
+ train_dataset=vectorized_datasets['train'] if training_args.do_train else None,
661
+ eval_dataset=vectorized_datasets['eval'] if training_args.do_eval else None,
662
+ data_collator=NeMoDataCollator,
663
+ )
664
+
665
+ # 8. Finally, we can start training
666
+
667
+ # Training
668
+ if training_args.do_train:
669
+
670
+ # use last checkpoint if exist
671
+ if last_checkpoint is not None:
672
+ checkpoint = last_checkpoint
673
+ elif model_args.model_name_or_path is not None and os.path.isdir(model_args.model_name_or_path):
674
+ checkpoint = model_args.model_name_or_path
675
+ else:
676
+ checkpoint = None
677
+
678
+ train_result = trainer.train(resume_from_checkpoint=checkpoint)
679
+ trainer.save_model()
680
+
681
+ metrics = train_result.metrics
682
+ max_train_samples = (
683
+ data_args.max_train_samples
684
+ if data_args.max_train_samples is not None
685
+ else len(vectorized_datasets["train"])
686
+ )
687
+ metrics["train_samples"] = min(max_train_samples, len(vectorized_datasets["train"]))
688
+
689
+ trainer.log_metrics("train", metrics)
690
+ trainer.save_metrics("train", metrics)
691
+ trainer.save_state()
692
+
693
+ # Change decoding strategy for final eval/predict
694
+ if training_args.do_eval or training_args.do_predict:
695
+ # set beam search decoding config
696
+ beam_decoding_config = copy.deepcopy(trainer.model.cfg.decoding)
697
+ beam_decoding_config.strategy = model_args.final_decoding_strategy
698
+ beam_decoding_config.beam.beam_size = model_args.final_num_beams
699
+
700
+ trainer.model.change_decoding_strategy(beam_decoding_config)
701
+
702
+ results = {}
703
+ if training_args.do_eval:
704
+ logger.info(f"*** Running Final Evaluation ({model_args.final_decoding_strategy}) ***")
705
+
706
+ predictions = trainer.transcribe(vectorized_datasets["eval"])
707
+ targets = model.tokenizer.ids_to_text(vectorized_datasets["eval"]["labels"])
708
+
709
+ cer_metric = load_metric("cer")
710
+ wer_metric = load_metric("wer")
711
+
712
+ cer = cer_metric.compute(predictions=predictions, references=targets)
713
+ wer = wer_metric.compute(predictions=predictions, references=targets)
714
+
715
+ metrics = {f"eval_cer": cer, f"eval_wer": wer}
716
+
717
+ max_eval_samples = (
718
+ data_args.max_eval_samples if data_args.max_eval_samples is not None else len(
719
+ vectorized_datasets["eval"])
720
+ )
721
+ metrics["eval_samples"] = min(max_eval_samples, len(vectorized_datasets["eval"]))
722
+
723
+ trainer.log_metrics("eval", metrics)
724
+ trainer.save_metrics("eval", metrics)
725
+
726
+ if "wandb" in training_args.report_to:
727
+ if not training_args.do_train:
728
+ wandb.init(name=training_args.run_name, project=data_args.wandb_project)
729
+ metrics = {os.path.join("eval", k[len("eval") + 1:]): v for k, v in metrics.items()}
730
+ # wandb.init(project=data_args.wandb_project, name=training_args.run_name)
731
+ wandb.log(metrics)
732
+ write_wandb_pred(predictions, targets, prefix="eval")
733
+
734
+ if training_args.do_predict:
735
+ logger.info(f"*** Running Final Prediction ({model_args.final_decoding_strategy}) ***")
736
+
737
+ for split in test_split:
738
+ predictions = trainer.transcribe(vectorized_datasets[split])
739
+ targets = model.tokenizer.ids_to_text(vectorized_datasets[split]["labels"])
740
+
741
+ cer_metric = load_metric("cer")
742
+ wer_metric = load_metric("wer")
743
+
744
+ cer = cer_metric.compute(predictions=predictions, references=targets)
745
+ wer = wer_metric.compute(predictions=predictions, references=targets)
746
+
747
+ metrics = {f"{split}_cer": cer, f"{split}_wer": wer}
748
+
749
+ max_predict_samples = (
750
+ data_args.max_predict_samples if data_args.max_predict_samples is not None else len(
751
+ vectorized_datasets[split])
752
+ )
753
+ metrics[f"{split}_samples"] = min(max_predict_samples, len(vectorized_datasets[split]))
754
+
755
+ trainer.log_metrics(split, metrics)
756
+ trainer.save_metrics(split, metrics)
757
+
758
+ if "wandb" in training_args.report_to:
759
+ if not training_args.do_train or training_args.do_eval:
760
+ wandb.init(name=training_args.run_name, project=data_args.wandb_project)
761
+ metrics = {os.path.join(split, k[len(split) + 1:]): v for k, v in metrics.items()}
762
+ wandb.log(metrics)
763
+ write_wandb_pred(predictions, targets, prefix=split)
764
+
765
+ # Write model card and (optionally) push to hub
766
+ config_name = data_args.dataset_config_name if data_args.dataset_config_name is not None else "na"
767
+ kwargs = {
768
+ "finetuned_from": model_args.model_name_or_path,
769
+ "tasks": "speech-recognition",
770
+ "tags": ["automatic-speech-recognition", data_args.dataset_name],
771
+ "dataset_args": (
772
+ f"Config: {config_name}, Training split: {data_args.train_split_name}, Eval split:"
773
+ f" {data_args.eval_split_name}"
774
+ ),
775
+ "dataset": f"{data_args.dataset_name.upper()} - {config_name.upper()}",
776
+ }
777
+ if "common_voice" in data_args.dataset_name:
778
+ kwargs["language"] = config_name
779
+
780
+ if training_args.push_to_hub:
781
+ trainer.push_to_hub(**kwargs)
782
+ #else:
783
+ #trainer.create_model_card(**kwargs)
784
+
785
+ return results
786
+
787
+
788
+ if __name__ == "__main__":
789
+ main()
run_spgispeech.sh ADDED
@@ -0,0 +1,32 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ #!/usr/bin/env bash
2
+ CUDA_VISIBLE_DEVICES=0 python run_speech_recognition_rnnt.py \
3
+ --config_path="conf/conformer_transducer_bpe_xlarge.yaml" \
4
+ --model_name_or_path="stt_en_conformer_transducer_xlarge" \
5
+ --dataset_name="esb/datasets" \
6
+ --tokenizer_path="tokenizer" \
7
+ --vocab_size="1024" \
8
+ --max_steps="100000" \
9
+ --dataset_config_name="spgispeech" \
10
+ --output_dir="./" \
11
+ --run_name="conformer-rnnt-spgispeech" \
12
+ --wandb_project="rnnt" \
13
+ --per_device_train_batch_size="8" \
14
+ --per_device_eval_batch_size="4" \
15
+ --logging_steps="50" \
16
+ --learning_rate="1e-4" \
17
+ --warmup_steps="500" \
18
+ --save_strategy="steps" \
19
+ --save_steps="20000" \
20
+ --evaluation_strategy="steps" \
21
+ --eval_steps="20000" \
22
+ --report_to="wandb" \
23
+ --preprocessing_num_workers="4" \
24
+ --fused_batch_size="4" \
25
+ --length_column_name="input_lengths" \
26
+ --fuse_loss_wer \
27
+ --group_by_length \
28
+ --overwrite_output_dir \
29
+ --do_train \
30
+ --do_eval \
31
+ --do_predict \
32
+ --use_auth_token
stt_en_conformer_transducer_xlarge.nemo ADDED
@@ -0,0 +1,3 @@
 
 
 
 
1
+ version https://git-lfs.github.com/spec/v1
2
+ oid sha256:6619304cc9b9924096b38299fc8ad959ff9ba079ac9cc980429a49d1cfc57d1b
3
+ size 2577971200