File size: 6,775 Bytes
dde077c
 
 
f8c0622
dde077c
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
0d4321a
dde077c
 
96e8d1e
dde077c
 
 
 
 
 
 
 
 
 
a5466cd
dde077c
a5466cd
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
dde077c
 
 
 
 
 
 
 
 
a5466cd
 
 
dde077c
 
ae187f9
dde077c
 
a5466cd
dde077c
 
 
 
ae187f9
dde077c
 
 
 
 
afba67b
 
dde077c
 
 
 
a5466cd
dde077c
 
74016c6
a5466cd
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
dde077c
 
 
 
 
 
 
402c7de
dde077c
 
 
 
 
a5466cd
 
 
 
dde077c
 
 
 
 
 
a5466cd
dde077c
a5466cd
 
dde077c
 
a5466cd
 
dde077c
 
 
 
 
 
0d4321a
dde077c
 
 
afba67b
 
 
 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
---
language: jv
datasets:
- openslr
metrics:
- wer
tags:
- audio
- automatic-speech-recognition
- speech
- xlsr-fine-tuning-week
license: apache-2.0
model-index:
- name: XLSR Wav2Vec2 Javanese by cahya
  results:
  - task: 
      name: Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: OpenSLR High quality TTS data for Javanese
      type: OpenSLR
      args: jv
    metrics:
       - name: Test WER
         type: wer
         value: 17.61
---

# Wav2Vec2-Large-XLSR-Javanese

Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53)
on the [OpenSLR High quality TTS data for Javanese](https://openslr.org/41/).
When using this model, make sure that your speech input is sampled at 16kHz.

## Usage
The model can be used directly (without a language model) as follows:
```python
import torch
import torchaudio
from datasets import load_dataset, load_metric, Dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
from datasets.utils.download_manager import DownloadManager
from pathlib import Path
import pandas as pd

def load_dataset_javanese():
    urls = [
        "https://www.openslr.org/resources/41/jv_id_female.zip",
        "https://www.openslr.org/resources/41/jv_id_male.zip"
    ]
    dm = DownloadManager()
    download_dirs = dm.download_and_extract(urls)
    data_dirs = [ 
        Path(download_dirs[0])/"jv_id_female/wavs",
        Path(download_dirs[1])/"jv_id_male/wavs",
    ]
    filenames = [ 
        Path(download_dirs[0])/"jv_id_female/line_index.tsv",
        Path(download_dirs[1])/"jv_id_male/line_index.tsv",
    ]
    
    dfs = []
    dfs.append(pd.read_csv(filenames[0], sep='\t', names=["path", "sentence"]))
    dfs.append(pd.read_csv(filenames[1], sep='\t', names=["path", "client_id", "sentence"]))
    dfs[1] = dfs[1].drop(["client_id"], axis=1)
    
    for i, dir in enumerate(data_dirs):
        dfs[i]["path"] = dfs[i].apply(lambda row: str(data_dirs[i]) + "/" + row + ".wav", axis=1)
    df = pd.concat(dfs)
    # df = df.sample(frac=1, random_state=1).reset_index(drop=True)
    dataset = Dataset.from_pandas(df)
    dataset = dataset.remove_columns('__index_level_0__')
    
    return dataset.train_test_split(test_size=0.1, seed=1)

dataset = load_dataset_javanese()
test_dataset = dataset['test']

processor = Wav2Vec2Processor.from_pretrained("cahya/wav2vec2-large-xlsr-javanese")
model = Wav2Vec2ForCTC.from_pretrained("cahya/wav2vec2-large-xlsr-javanese")

resampler = torchaudio.transforms.Resample(48_000, 16_000)

# Preprocessing the datasets.
# We need to read the aduio files as arrays
def speech_file_to_array_fn(batch):
    speech_array, sampling_rate = torchaudio.load(batch["path"])
    batch["speech"] = resampler(speech_array).squeeze().numpy()
    return batch

test_dataset = test_dataset.map(speech_file_to_array_fn)
inputs = processor(test_dataset[:2]["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)

with torch.no_grad():
    logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits

predicted_ids = torch.argmax(logits, dim=-1)

print("Prediction:", processor.batch_decode(predicted_ids))
print("Reference:", test_dataset[:2]["sentence"])
```


## Evaluation

The model can be evaluated as follows or using this
[notebook](https://github.com/cahya-wirawan/indonesian-speech-recognition/blob/main/XLSR_Wav2Vec2_for_Indonesian_Evaluation-Javanese.ipynb)

```python
import torch
import torchaudio
from datasets import load_dataset, load_metric, Dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re
from datasets.utils.download_manager import DownloadManager
from pathlib import Path
import pandas as pd

def load_dataset_javanese():
    urls = [
        "https://www.openslr.org/resources/41/jv_id_female.zip",
        "https://www.openslr.org/resources/41/jv_id_male.zip"
    ]
    dm = DownloadManager()
    download_dirs = dm.download_and_extract(urls)
    data_dirs = [
        Path(download_dirs[0])/"jv_id_female/wavs",
        Path(download_dirs[1])/"jv_id_male/wavs",
    ]
    filenames = [
        Path(download_dirs[0])/"jv_id_female/line_index.tsv",
        Path(download_dirs[1])/"jv_id_male/line_index.tsv",
    ]

    dfs = []
    dfs.append(pd.read_csv(filenames[0], sep='\t', names=["path", "sentence"]))
    dfs.append(pd.read_csv(filenames[1], sep='\t', names=["path", "client_id", "sentence"]))
    dfs[1] = dfs[1].drop(["client_id"], axis=1)

    for i, dir in enumerate(data_dirs):
        dfs[i]["path"] = dfs[i].apply(lambda row: str(data_dirs[i]) + "/" + row + ".wav", axis=1)
    df = pd.concat(dfs)
    # df = df.sample(frac=1, random_state=1).reset_index(drop=True)
    dataset = Dataset.from_pandas(df)
    dataset = dataset.remove_columns('__index_level_0__')

    return dataset.train_test_split(test_size=0.1, seed=1)

dataset = load_dataset_javanese()
test_dataset = dataset['test']

wer = load_metric("wer")

processor = Wav2Vec2Processor.from_pretrained("cahya/wav2vec2-large-xlsr-javanese")
model = Wav2Vec2ForCTC.from_pretrained("cahya/wav2vec2-large-xlsr-javanese") 
model.to("cuda")

chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"\“\%\‘\'\”_\�]'
resampler = torchaudio.transforms.Resample(48_000, 16_000)

# Preprocessing the datasets.
# We need to read the aduio files as arrays
def speech_file_to_array_fn(batch):
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
    speech_array, sampling_rate = torchaudio.load(batch["path"])
    batch["speech"] = resampler(speech_array).squeeze().numpy()
    return batch

test_dataset = test_dataset.map(speech_file_to_array_fn)

# Preprocessing the datasets.
# We need to read the aduio files as arrays
def evaluate(batch):
    inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)

    with torch.no_grad():
        logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits

    pred_ids = torch.argmax(logits, dim=-1)
    batch["pred_strings"] = processor.batch_decode(pred_ids)
    return batch

result = test_dataset.map(evaluate, batched=True, batch_size=8)

print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
```

**Test Result**: 17.61 %

## Training

[OpenSLR High quality TTS data for Javanese](https://openslr.org/41/) was used for training.
The script used for training can be found [here](https://github.com/cahya-wirawan/indonesian-speech-recognition/blob/main/XLSR_Wav2Vec2_for_Indonesian_Evaluation-Javanese.ipynb) 
and to [evaluate it](https://github.com/cahya-wirawan/indonesian-speech-recognition/blob/main/XLSR_Wav2Vec2_for_Indonesian_Evaluation-Javanese.ipynb)