File size: 9,407 Bytes
93533d4
 
 
 
 
 
 
fd13715
6645145
93533d4
fd13715
202a8b5
fd13715
 
 
 
 
 
 
 
 
 
 
c102c06
93533d4
 
c102c06
202a8b5
c102c06
 
 
93533d4
 
 
 
 
 
 
 
61e51f4
 
93533d4
 
 
 
 
61e51f4
93533d4
61e51f4
93533d4
202a8b5
93533d4
 
 
 
61e51f4
 
 
93533d4
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
61e51f4
 
 
 
 
 
 
 
 
 
 
 
93533d4
 
 
 
 
 
 
c102c06
 
 
 
93533d4
c102c06
 
 
93533d4
 
 
 
 
 
 
 
c102c06
93533d4
c102c06
93533d4
c102c06
93533d4
202a8b5
93533d4
 
 
 
3141b41
93533d4
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
c102c06
 
93533d4
 
c102c06
93533d4
c102c06
93533d4
c102c06
 
 
 
93533d4
 
 
 
 
 
 
 
c102c06
93533d4
 
c102c06
93533d4
c102c06
202a8b5
93533d4
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
c102c06
 
93533d4
 
c102c06
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
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
---
language: zh
datasets:
- common_voice
tags:
- audio
- automatic-speech-recognition
- speech
- xlsr-fine-tuning-week
license: apache-2.0
model-index:
- name: XLSR Wav2Vec2 Taiwanese Mandarin(zh-tw) by Voidful
  results:
  - task: 
      name: Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: Common Voice zh-TW
      type: common_voice
      args: zh-TW 
    metrics:
       - name: Test CER
         type: cer
         value: 16.41
---

# Wav2Vec2-Large-XLSR-53-tw-gpt
Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on zh-tw using the [Common Voice](https://huggingface.co/datasets/common_voice).   
When using this model, make sure that your speech input is sampled at 16kHz.

## Usage
[Colab trial](https://colab.research.google.com/drive/1e_z5jQHYbO2YKEaUgzb1ww1WwiAyydAj?usp=sharing)

```
import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
    AutoTokenizer, 
    AutoModelWithLMHead 
)
import torch
import re
import sys

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"

chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\\\\\\\\"#$%&()*+,\\\\\\\\-.\\\\\\\\:;<=>?@\\\\\\\\[\\\\\\\\]\\\\\\\\\\\\\\\\\\\\\\\\/^_`{|}~]"

model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

tokenizer = AutoTokenizer.from_pretrained("ckiplab/gpt2-base-chinese")  
gpt_model = AutoModelWithLMHead.from_pretrained("ckiplab/gpt2-base-chinese").to(device)

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

def load_file_to_data(file):
    batch = {}
    speech, _ = torchaudio.load(file)
    batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
    batch["sampling_rate"] = resampler.new_freq
    return batch

def predict(data):
    features = processor(data["speech"], sampling_rate=data["sampling_rate"], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits
    
    decoded_results = []
    for logit in logits:
        pred_ids = torch.argmax(logit, dim=-1)
        mask = pred_ids.ge(1).unsqueeze(-1).expand(logit.size())
        vocab_size = logit.size()[-1]
        voice_prob = torch.nn.functional.softmax((torch.masked_select(logit, mask).view(-1,vocab_size)),dim=-1)
        gpt_input = torch.cat((torch.tensor([tokenizer.cls_token_id]).to(device),pred_ids[pred_ids>0]), 0)
        gpt_prob = torch.nn.functional.softmax(gpt_model(gpt_input).logits, dim=-1)[:voice_prob.size()[0],:]
        comb_pred_ids = torch.argmax(gpt_prob*voice_prob, dim=-1)
        decoded_results.append(processor.decode(comb_pred_ids))

    return decoded_results
```

Predict
```python
predict(load_file_to_data('voice file path'))
```

## Evaluation
The model can be evaluated as follows on the zh-tw test data of Common Voice.   
CER calculation refer to https://huggingface.co/ctl/wav2vec2-large-xlsr-cantonese 

```python
!mkdir cer
!pip install jiwer

import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"

chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\\\\\\\\"#$%&()*+,\\\\\\\\-.\\\\\\\\:;<=>?@\\\\\\\\[\\\\\\\\]\\\\\\\\\\\\\\\\\\\\\\\\/^_`{|}~]"

model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

ds = load_dataset("common_voice", 'zh-TW', split="test")

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

def map_to_array(batch):
    speech, _ = torchaudio.load(batch["path"])
    batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
    batch["sampling_rate"] = resampler.new_freq
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch

ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits
    pred_ids = torch.argmax(logits, dim=-1)
    batch["predicted"] = processor.batch_decode(pred_ids)
    batch["target"] = batch["sentence"]
    return batch

result = ds.map(map_to_pred, batched=True, batch_size=16, remove_columns=list(ds.features.keys()))

cer = load_metric("./cer")
print("CER: {:2f}".format(100 * cer.compute(predictions=result["predicted"], references=result["target"])))
```

`CER: 28.734822`

## Evaluation with GPT:
```python
!mkdir cer
!wget -O cer/cer.py https://huggingface.co/ctl/wav2vec2-large-xlsr-cantonese/raw/main/cer.py
!pip install jiwer

import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys
from transformers import AutoTokenizer, AutoModelWithLMHead 

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
chars_to_ignore_regex = r"""[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\\\\\\\\"#$%&()*+,\\\\\\\\-.\\\\\\\\:;<=>?@\\\\\\\\[\\\\\\\\]\\\\\\\\\\\\\\\\\\\\\\\\/^_`{|}~]"""

tokenizer = AutoTokenizer.from_pretrained("ckiplab/gpt2-base-chinese")  
gpt_model = AutoModelWithLMHead.from_pretrained("ckiplab/gpt2-base-chinese").to(device)
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

ds = load_dataset("common_voice", 'zh-TW', data_dir="./cv-corpus-6.1-2020-12-11", split="test")

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

def map_to_array(batch):
    speech, _ = torchaudio.load(batch["path"])
    batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
    batch["sampling_rate"] = resampler.new_freq
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch

ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits
    
    decoded_results = []
    for logit in logits:
        pred_ids = torch.argmax(logit, dim=-1)
        mask = pred_ids.ge(1).unsqueeze(-1).expand(logit.size())
        vocab_size = logit.size()[-1]
        voice_prob = torch.nn.functional.softmax((torch.masked_select(logit, mask).view(-1,vocab_size)),dim=-1)
        gpt_input = torch.cat((torch.tensor([tokenizer.cls_token_id]).to(device),pred_ids[pred_ids>0]), 0)
        gpt_prob = torch.nn.functional.softmax(gpt_model(gpt_input).logits, dim=-1)[:voice_prob.size()[0],:]
        comb_pred_ids = torch.argmax(gpt_prob*voice_prob, dim=-1)
        decoded_results.append(processor.decode(comb_pred_ids))
        
    batch["predicted"] = decoded_results
    batch["target"] = batch["sentence"]
    return batch


result = ds.map(map_to_pred, batched=True, batch_size=16, remove_columns=list(ds.features.keys()))

cer = load_metric("./cer")
print("CER: {:2f}".format(100 * cer.compute(predictions=result["predicted"], references=result["target"])))
```

`CER 25.69`